QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track M. Thomson, Ed.
Expires: November 23, 2018 Mozilla
May 22, 2018
QUIC: A UDP-Based Multiplexed and Secure Transportdraft-ietf-quic-transport-12
Abstract
This document defines the core of the QUIC transport protocol. This
document describes connection establishment, packet format,
multiplexing and reliability. Accompanying documents describe the
cryptographic handshake and loss detection.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
<https://mailarchive.ietf.org/arch/search/?email_list=quic>.
Working Group information can be found at <https://github.com/quicwg>; source code and issues list for this draft can be found at
<https://github.com/quicwg/base-drafts/labels/-transport>.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 23, 2018.
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QUIC version 1 conforms to the protocol invariants in
[QUIC-INVARIANTS].
2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
Definitions of terms that are used in this document:
Client: The endpoint initiating a QUIC connection.
Server: The endpoint accepting incoming QUIC connections.
Endpoint: The client or server end of a connection.
Stream: A logical, bi-directional channel of ordered bytes within a
QUIC connection.
Connection: A conversation between two QUIC endpoints with a single
encryption context that multiplexes streams within it.
Connection ID: An opaque identifier that is used to identify a QUIC
connection at an endpoint. Each endpoint sets a value that its
peer includes in packets.
QUIC packet: A well-formed UDP payload that can be parsed by a QUIC
receiver.
QUIC is a name, not an acronym.
2.1. Notational Conventions
Packet and frame diagrams use the format described in Section 3.1 of
[RFC2360], with the following additional conventions:
[x] Indicates that x is optional
x (A) Indicates that x is A bits long
x (A/B/C) ... Indicates that x is one of A, B, or C bits long
x (i) ... Indicates that x uses the variable-length encoding in
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x (*) ... Indicates that x is variable-length
3. Versions
QUIC versions are identified using a 32-bit unsigned number.
The version 0x00000000 is reserved to represent version negotiation.
This version of the specification is identified by the number
0x00000001.
Other versions of QUIC might have different properties to this
version. The properties of QUIC that are guaranteed to be consistent
across all versions of the protocol are described in
[QUIC-INVARIANTS].
Version 0x00000001 of QUIC uses TLS as a cryptographic handshake
protocol, as described in [QUIC-TLS].
Versions with the most significant 16 bits of the version number
cleared are reserved for use in future IETF consensus documents.
Versions that follow the pattern 0x?a?a?a?a are reserved for use in
forcing version negotiation to be exercised. That is, any version
number where the low four bits of all octets is 1010 (in binary). A
client or server MAY advertise support for any of these reserved
versions.
Reserved version numbers will probably never represent a real
protocol; a client MAY use one of these version numbers with the
expectation that the server will initiate version negotiation; a
server MAY advertise support for one of these versions and can expect
that clients ignore the value.
[[RFC editor: please remove the remainder of this section before
publication.]]
The version number for the final version of this specification
(0x00000001), is reserved for the version of the protocol that is
published as an RFC.
Version numbers used to identify IETF drafts are created by adding
the draft number to 0xff000000. For example, draft-ietf-quic-transport-13 would be identified as 0xff00000D.
Implementors are encouraged to register version numbers of QUIC that
they are using for private experimentation on the GitHub wiki at
<https://github.com/quicwg/base-drafts/wiki/QUIC-Versions>.
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We first describe QUIC's packet types and their formats, since some
are referenced in subsequent mechanisms.
All numeric values are encoded in network byte order (that is, big-
endian) and all field sizes are in bits. When discussing individual
bits of fields, the least significant bit is referred to as bit 0.
Hexadecimal notation is used for describing the value of fields.
Any QUIC packet has either a long or a short header, as indicated by
the Header Form bit. Long headers are expected to be used early in
the connection before version negotiation and establishment of 1-RTT
keys. Short headers are minimal version-specific headers, which are
used after version negotiation and 1-RTT keys are established.
4.1. Long Header
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1| Type (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Long Header Format
Long headers are used for packets that are sent prior to the
completion of version negotiation and establishment of 1-RTT keys.
Once both conditions are met, a sender switches to sending packets
using the short header (Section 4.2). The long form allows for
special packets - such as the Version Negotiation packet - to be
represented in this uniform fixed-length packet format. A long
header contains the following fields:
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Header Form: The most significant bit (0x80) of octet 0 (the first
octet) is set to 1 for long headers.
Long Packet Type: The remaining seven bits of octet 0 contain the
packet type. This field can indicate one of 128 packet types.
The types specified for this version are listed in Table 1.
Version: The QUIC Version is a 32-bit field that follows the Type.
This field indicates which version of QUIC is in use and
determines how the rest of the protocol fields are interpreted.
DCIL and SCIL: Octet 1 contains the lengths of the two connection ID
fields that follow it. These lengths are encoded as two 4-bit
unsigned integers. The Destination Connection ID Length (DCIL)
field occupies the 4 high bits of the octet and the Source
Connection ID Length (SCIL) field occupies the 4 low bits of the
octet. An encoded length of 0 indicates that the connection ID is
also 0 octets in length. Non-zero encoded lengths are increased
by 3 to get the full length of the connection ID, producing a
length between 4 and 18 octets inclusive. For example, an octet
with the value 0x50 describes an 8-octet Destination Connection ID
and a zero-length Source Connection ID.
Destination Connection ID: The Destination Connection ID field
follows the connection ID lengths and is either 0 octets in length
or between 4 and 18 octets. Section 4.7 describes the use of this
field in more detail.
Source Connection ID: The Source Connection ID field follows the
Destination Connection ID and is either 0 octets in length or
between 4 and 18 octets. Section 4.7 describes the use of this
field in more detail.
Payload Length: The length of the Payload field in octets, encoded
as a variable-length integer (Section 7.1).
Packet Number: The packet number field is 1, 2, or 4 octets long.
The packet number has confidentiality protection separate from
packet protection, as described in Section 5.6 of [QUIC-TLS]. The
length of the packet number field is encoded in the plaintext
packet number. See Section 4.8 for details.
Payload: The payload of the packet.
The following packet types are defined:
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Header Form: The most significant bit (0x80) of octet 0 is set to 0
for the short header.
Key Phase Bit: The second bit (0x40) of octet 0 indicates the key
phase, which allows a recipient of a packet to identify the packet
protection keys that are used to protect the packet. See
[QUIC-TLS] for details.
[[Editor's Note: this section should be removed and the bit
definitions changed before this draft goes to the IESG.]]
Third Bit: The third bit (0x20) of octet 0 is set to 1.
[[Editor's Note: this section should be removed and the bit
definitions changed before this draft goes to the IESG.]]
Fourth Bit: The fourth bit (0x10) of octet 0 is set to 1.
[[Editor's Note: this section should be removed and the bit
definitions changed before this draft goes to the IESG.]]
Google QUIC Demultipexing Bit: The fifth bit (0x8) of octet 0 is set
to 0. This allows implementations of Google QUIC to distinguish
Google QUIC packets from short header packets sent by a client
because Google QUIC servers expect the connection ID to always be
present. The special interpretation of this bit SHOULD be removed
from this specification when Google QUIC has finished
transitioning to the new header format.
Reserved: The sixth, seventh, and eighth bits (0x7) of octet 0 are
reserved for experimentation.
Destination Connection ID: The Destination Connection ID is a
connection ID that is chosen by the intended recipient of the
packet. See Section 4.7 for more details.
Packet Number: The packet number field is 1, 2, or 4 octets long.
The packet number has confidentiality protection separate from
packet protection, as described in Section 5.6 of [QUIC-TLS]. The
length of the packet number field is encoded in the plaintext
packet number. See Section 4.8 for details.
Protected Payload: Packets with a short header always include a
1-RTT protected payload.
The packet type in a short header currently determines only the size
of the packet number field. Additional types can be used to signal
the presence of other fields.
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The header form and connection ID field of a short header packet are
version-independent. The remaining fields are specific to the
selected QUIC version. See [QUIC-INVARIANTS] for details on how
packets from different versions of QUIC are interpreted.
4.3. Version Negotiation Packet
A Version Negotiation packet is inherently not version-specific, and
does not use the long packet header (see Section 4.1. Upon receipt
by a client, it will appear to be a packet using the long header, but
will be identified as a Version Negotiation packet based on the
Version field having a value of 0.
The Version Negotiation packet is a response to a client packet that
contains a version that is not supported by the server, and is only
sent by servers.
The layout of a Version Negotiation packet is:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1| Unused (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Supported Version 1 (32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version 2 (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version N (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Version Negotiation Packet
The value in the Unused field is selected randomly by the server.
The Version field of a Version Negotiation packet MUST be set to
0x00000000.
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The server MUST include the value from the Source Connection ID field
of the packet it receives in the Destination Connection ID field.
The value for Source Connection ID MUST be copied from the
Destination Connection ID of the received packet, which is initially
randomly selected by a client. Echoing both connection IDs gives
clients some assurance that the server received the packet and that
the Version Negotiation packet was not generated by an off-path
attacker.
The remainder of the Version Negotiation packet is a list of 32-bit
versions which the server supports.
A Version Negotiation packet cannot be explicitly acknowledged in an
ACK frame by a client. Receiving another Initial packet implicitly
acknowledges a Version Negotiation packet.
The Version Negotiation packet does not include the Packet Number and
Length fields present in other packets that use the long header form.
Consequently, a Version Negotiation packet consumes an entire UDP
datagram.
See Section 6.2 for a description of the version negotiation process.
4.4. Cryptographic Handshake Packets
Once version negotiation is complete, the cryptographic handshake is
used to agree on cryptographic keys. The cryptographic handshake is
carried in Initial (Section 4.4.1), Retry (Section 4.4.2) and
Handshake (Section 4.4.3) packets.
All these packets use the long header and contain the current QUIC
version in the version field.
In order to prevent tampering by version-unaware middleboxes,
handshake packets are protected with a connection- and version-
specific key, as described in [QUIC-TLS]. This protection does not
provide confidentiality or integrity against on-path attackers, but
provides some level of protection against off-path attackers.
4.4.1. Initial Packet
The Initial packet uses long headers with a type value of 0x7F. It
carries the first cryptographic handshake message sent by the client.
If the client has not previously received a Retry packet from the
server, it populates the Destination Connection ID field with a
randomly selected value. This MUST be at least 8 octets in length.
Until a packet is received from the server, the client MUST use the
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same random value unless it also changes the Source Connection ID
(which effectively starts a new connection attempt). The randomized
Destination Connection ID is used to determine packet protection
keys.
If the client received a Retry packet and is sending a second Initial
packet, then it sets the Destination Connection ID to the value from
the Source Connection ID in the Retry packet. Changing Destination
Connection ID also results in a change to the keys used to protect
the Initial packet.
The client populates the Source Connection ID field with a value of
its choosing and sets the SCIL field to match.
The first Initial packet that is sent by a client contains a packet
number of 0. All subsequent packets contain a packet number that is
incremented by at least one, see (Section 4.8).
The payload of an Initial packet conveys a STREAM frame (or frames)
for stream 0 containing a cryptographic handshake message. The
stream in this packet always starts at an offset of 0 (see
Section 6.5) and the complete cryptographic handshake message MUST
fit in a single packet (see Section 6.3).
The payload of a UDP datagram carrying the Initial packet MUST be
expanded to at least 1200 octets (see Section 8), by adding PADDING
frames to the Initial packet and/or by combining the Initial packet
with a 0-RTT packet (see Section 4.6).
The client uses the Initial packet type for any packet that contains
an initial cryptographic handshake message. This includes all cases
where a new packet containing the initial cryptographic message needs
to be created, this includes the packets sent after receiving a
Version Negotiation (Section 4.3) or Retry packet (Section 4.4.2).
4.4.2. Retry Packet
A Retry packet uses long headers with a type value of 0x7E. It
carries cryptographic handshake messages and acknowledgments. It is
used by a server that wishes to perform a stateless retry (see
Section 6.5).
The server populates the Destination Connection ID with the
connection ID that the client included in the Source Connection ID of
the Initial packet. This might be a zero-length value.
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The server includes a connection ID of its choice in the Source
Connection ID field. The client MUST use this connection ID in the
Destination Connection ID of subsequent packets that it sends.
The Packet Number field of a Retry packet MUST be set to 0. This
value is subsequently protected as normal. [[Editor's Note: This
isn't ideal, because it creates a "cheat" where the client assumes a
value. That's a problem, so I'm tempted to suggest that this include
any value less than 2^30 so that normal processing works - and can be
properly exercised.]]
A Retry packet is never explicitly acknowledged in an ACK frame by a
client. Receiving another Initial packet implicitly acknowledges a
Retry packet.
After receiving a Retry packet, the client uses a new Initial packet
containing the next cryptographic handshake message. The client
retains the state of its cryptographic handshake, but discards all
transport state. The Initial packet that is generated in response to
a Retry packet includes STREAM frames on stream 0 that start again at
an offset of 0.
Continuing the cryptographic handshake is necessary to ensure that an
attacker cannot force a downgrade of any cryptographic parameters.
In addition to continuing the cryptographic handshake, the client
MUST remember the results of any version negotiation that occurred
(see Section 6.2). The client MAY also retain any observed RTT or
congestion state that it has accumulated for the flow, but other
transport state MUST be discarded.
The payload of the Retry packet contains at least two frames. It
MUST include a STREAM frame on stream 0 with offset 0 containing the
server's cryptographic stateless retry material. It MUST also
include an ACK frame to acknowledge the client's Initial packet. It
MAY additionally include PADDING frames. The next STREAM frame sent
by the server will also start at stream offset 0.
4.4.3. Handshake Packet
A Handshake packet uses long headers with a type value of 0x7D. It
is used to carry acknowledgments and cryptographic handshake messages
from the server and client.
A server sends its cryptographic handshake in one or more Handshake
packets in response to an Initial packet if it does not send a Retry
packet. Once a client has received a Handshake packet from a server,
it uses Handshake packets to send subsequent cryptographic handshake
messages and acknowledgments to the server.
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The Destination Connection ID field in a Handshake packet contains a
connection ID that is chosen by the recipient of the packet; the
Source Connection ID includes the connection ID that the sender of
the packet wishes to use (see Section 4.7).
The first Handshake packet sent by a server contains a packet number
of 0. Packet numbers are incremented normally for other Handshake
packets.
Servers MUST NOT send more than three Handshake packets without
receiving a packet from a verified source address. Source addresses
can be verified through an address validation token, receipt of the
final cryptographic message from the client, or by receiving a valid
PATH_RESPONSE frame from the client.
If the server expects to generate more than three Handshake packets
in response to an Initial packet, it SHOULD include a PATH_CHALLENGE
frame in each Handshake packet that it sends. After receiving at
least one valid PATH_RESPONSE frame, the server can send its
remaining Handshake packets. Servers can instead perform address
validation using a Retry packet; this requires less state on the
server, but could involve additional computational effort depending
on implementation choices.
The payload of this packet contains STREAM frames and could contain
PADDING, ACK, PATH_CHALLENGE, or PATH_RESPONSE frames. Handshake
packets MAY contain CONNECTION_CLOSE frames if the handshake is
unsuccessful.
4.5. Protected Packets
All QUIC packets are protected. Packets that are protected with the
static handshake keys or the 0-RTT keys are sent with long headers;
all packets protected with 1-RTT keys are sent with short headers.
The different packet types explicitly indicate the encryption level
and therefore the keys that are used to remove packet protection.
Packets protected with 0-RTT keys use a type value of 0x7C. The
connection ID fields for a 0-RTT packet MUST match the values used in
the Initial packet (Section 4.4.1).
The client can send 0-RTT packets after receiving a Handshake packet
(Section 4.4.3), if that packet does not complete the handshake.
Even if the client receives a different connection ID in the
Handshake packet, it MUST continue to use the same Destination
Connection ID for 0-RTT packets, see Section 4.7.
The version field for protected packets is the current QUIC version.
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The packet number field contains a packet number, which has
additional confidentiality protection that is applied after packet
protection is applied (see [QUIC-TLS] for details). The underlying
packet number increases with each packet sent, see Section 4.8 for
details.
The payload is protected using authenticated encryption. [QUIC-TLS]
describes packet protection in detail. After decryption, the
plaintext consists of a sequence of frames, as described in
Section 5.
4.6. Coalescing Packets
A sender can coalesce multiple QUIC packets (typically a
Cryptographic Handshake packet and a Protected packet) into one UDP
datagram. This can reduce the number of UDP datagrams needed to send
application data during the handshake and immediately afterwards. A
packet with a short header does not include a length, so it has to be
the last packet included in a UDP datagram.
The sender MUST NOT coalesce QUIC packets belonging to different QUIC
connections into a single UDP datagram.
Every QUIC packet that is coalesced into a single UDP datagram is
separate and complete. Though the values of some fields in the
packet header might be redundant, no fields are omitted. The
receiver of coalesced QUIC packets MUST individually process each
QUIC packet and separately acknowledge them, as if they were received
as the payload of different UDP datagrams.
4.7. Connection ID
A connection ID is used to ensure consistent routing of packets. The
long header contains two connection IDs: the Destination Connection
ID is chosen by the recipient of the packet and is used to provide
consistent routing; the Source Connection ID is used to set the
Destination Connection ID used by the peer.
During the handshake, packets with the long header are used to
establish the connection ID that each endpoint uses. Each endpoint
uses the Source Connection ID field to specify the connection ID that
is used in the Destination Connection ID field of packets being sent
to them. Upon receiving a packet, each endpoint sets the Destination
Connection ID it sends to match the value of the Source Connection ID
that they receive.
During the handshake, an endpoint might receive multiple packets with
the long header, and thus be given multiple opportunities to update
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the Destination Connection ID it sends. A client MUST only change
the value it sends in the Destination Connection ID in response to
the first packet of each type it receives from the server (Retry or
Handshake); a server MUST set its value based on the Initial packet.
Any additional changes are not permitted; if subsequent packets of
those types include a different Source Connection ID, they MUST be
discarded. This avoids problems that might arise from stateless
processing of multiple Initial packets producing different connection
IDs.
Short headers only include the Destination Connection ID and omit the
explicit length. The length of the Destination Connection ID field
is expected to be known to endpoints.
Endpoints using a connection-ID based load balancer could agree with
the load balancer on a fixed or minimum length and on an encoding for
connection IDs. This fixed portion could encode an explicit length,
which allows the entire connection ID to vary in length and still be
used by the load balancer.
The very first packet sent by a client includes a random value for
Destination Connection ID. The same value MUST be used for all 0-RTT
packets sent on that connection (Section 4.5). This randomized value
is used to determine the handshake packet protection keys (see
Section 5.3.2 of [QUIC-TLS]).
A Version Negotiation (Section 4.3) packet MUST use both connection
IDs selected by the client, swapped to ensure correct routing toward
the client.
The connection ID can change over the lifetime of a connection,
especially in response to connection migration (Section 6.8).
NEW_CONNECTION_ID frames (Section 7.13) are used to provide new
connection ID values.
4.8. Packet Numbers
The packet number is an integer in the range 0 to 2^62-1. The value
is used in determining the cryptographic nonce for packet encryption.
Each endpoint maintains a separate packet number for sending and
receiving. The packet number for sending MUST start at zero for the
first packet sent and MUST increase by at least one after sending a
packet.
A QUIC endpoint MUST NOT reuse a packet number within the same
connection (that is, under the same cryptographic keys). If the
packet number for sending reaches 2^62 - 1, the sender MUST close the
connection without sending a CONNECTION_CLOSE frame or any further
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packets; a server MAY send a Stateless Reset (Section 6.10.4) in
response to further packets that it receives.
In the QUIC long and short packet headers, the number of bits
required to represent the packet number are reduced by including only
a variable number of the least significant bits of the packet number.
One or two of the most significant bits of the first octet determine
how many bits of the packet number are provided, as shown in Table 2.
+---------------------+----------------+--------------+
| First octet pattern | Encoded Length | Bits Present |
+---------------------+----------------+--------------+
| 0b0xxxxxxx | 1 octet | 7 |
| | | |
| 0b10xxxxxx | 2 | 14 |
| | | |
| 0b11xxxxxx | 4 | 30 |
+---------------------+----------------+--------------+
Table 2: Packet Number Encodings for Packet Headers
Note that these encodings are similar to those in Section 7.1, but
use different values.
The encoded packet number is protected as described in Section 5.6
[QUIC-TLS]. Protection of the packet number is removed prior to
recovering the full packet number. The full packet number is
reconstructed at the receiver based on the number of significant bits
present, the content of those bits, and the largest packet number
received on a successfully authenticated packet. Recovering the full
packet number is necessary to successfully remove packet protection.
Once packet number protection is removed, the packet number is
decoded by finding the packet number value that is closest to the
next expected packet. The next expected packet is the highest
received packet number plus one. For example, if the highest
successfully authenticated packet had a packet number of 0xaa82f30e,
then a packet containing a 14-bit value of 0x1f94 will be decoded as
0xaa831f94.
The sender MUST use a packet number size able to represent more than
twice as large a range than the difference between the largest
acknowledged packet and packet number being sent. A peer receiving
the packet will then correctly decode the packet number, unless the
packet is delayed in transit such that it arrives after many higher-
numbered packets have been received. An endpoint SHOULD use a large
enough packet number encoding to allow the packet number to be
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recovered even if the packet arrives after packets that are sent
afterwards.
As a result, the size of the packet number encoding is at least one
more than the base 2 logarithm of the number of contiguous
unacknowledged packet numbers, including the new packet.
For example, if an endpoint has received an acknowledgment for packet
0x6afa2f, sending a packet with a number of 0x6b2d79 requires a
packet number encoding with 14 bits or more; whereas the 30-bit
packet number encoding is needed to send a packet with a number of
0x6bc107.
A Version Negotiation packet (Section 4.3) does not include a packet
number. The Retry packet (Section 4.4.2) has special rules for
populating the packet number field.
5. Frames and Frame Types
The payload of all packets, after removing packet protection,
consists of a sequence of frames, as shown in Figure 4. Version
Negotiation and Stateless Reset do not contain frames.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 1 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 2 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame N (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: Contents of Protected Payload
Protected payloads MUST contain at least one frame, and MAY contain
multiple frames and multiple frame types.
Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC
packet boundary. Each frame begins with a Frame Type byte,
indicating its type, followed by additional type-dependent fields:
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Internet-Draft QUIC Transport Protocol May 2018Section 6.8. Finally a connection may be terminated by either
endpoint, as described in Section 6.10.
6.1. Matching Packets to Connections
Incoming packets are classified on receipt. Packets can either be
associated with an existing connection, or - for servers -
potentially create a new connection.
Hosts try to associate a packet with an existing connection. If the
packet has a Destination Connection ID corresponding to an existing
connection, QUIC processes that packet accordingly. Note that a
NEW_CONNECTION_ID frame (Section 7.13) would associate more than one
connection ID with a connection.
If the Destination Connection ID is zero length and the packet
matches the address/port tuple of a connection where the host did not
require connection IDs, QUIC processes the packet as part of that
connection. Endpoints MUST drop packets with zero-length Destination
Connection ID fields if they do not correspond to a single
connection.
6.1.1. Client Packet Handling
Valid packets sent to clients always include a Destination Connection
ID that matches the value the client selects. Clients that choose to
receive zero-length connection IDs can use the address/port tuple to
identify a connection. Packets that don't match an existing
connection MAY be discarded.
Due to packet reordering or loss, clients might receive packets for a
connection that are encrypted with a key it has not yet computed.
Clients MAY drop these packets, or MAY buffer them in anticipation of
later packets that allow it to compute the key.
If a client receives a packet that has an unsupported version, it
MUST discard that packet.
6.1.2. Server Packet Handling
If a server receives a packet that has an unsupported version and
sufficient length to be an Initial packet for some version supported
by the server, it SHOULD send a Version Negotiation packet as
described in Section 6.2.1. Servers MAY rate control these packets
to avoid storms of Version Negotiation packets.
The first packet for an unsupported version can use different
semantics and encodings for any version-specific field. In
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particular, different packet protection keys might be used for
different versions. Servers that do not support a particular version
are unlikely to be able to decrypt the content of the packet.
Servers SHOULD NOT attempt to decode or decrypt a packet from an
unknown version, but instead send a Version Negotiation packet,
provided that the packet is sufficiently long.
Servers MUST drop other packets that contain unsupported versions.
Packets with a supported version, or no version field, are matched to
a connection as described in Section 6.1. If not matched, the server
continues below.
If the packet is an Initial packet fully conforming with the
specification, the server proceeds with the handshake (Section 6.3).
This commits the server to the version that the client selected.
If a server isn't currently accepting any new connections, it SHOULD
send a Handshake packet containing a CONNECTION_CLOSE frame with
error code SERVER_BUSY.
If the packet is a 0-RTT packet, the server MAY buffer a limited
number of these packets in anticipation of a late-arriving Initial
Packet. Clients are forbidden from sending Handshake packets prior
to receiving a server response, so servers SHOULD ignore any such
packets.
Servers MUST drop incoming packets under all other circumstances.
They SHOULD send a Stateless Reset (Section 6.10.4) if a connection
ID is present in the header.
6.2. Version Negotiation
Version negotiation ensures that client and server agree to a QUIC
version that is mutually supported. A server sends a Version
Negotiation packet in response to each packet that might initiate a
new connection, see Section 6.1 for details.
The size of the first packet sent by a client will determine whether
a server sends a Version Negotiation packet. Clients that support
multiple QUIC versions SHOULD pad their Initial packets to reflect
the largest minimum Initial packet size of all their versions. This
ensures that that the server responds if there are any mutually
supported versions.
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Internet-Draft QUIC Transport Protocol May 20186.2.1. Sending Version Negotiation Packets
If the version selected by the client is not acceptable to the
server, the server responds with a Version Negotiation packet (see
Section 4.3). This includes a list of versions that the server will
accept.
This system allows a server to process packets with unsupported
versions without retaining state. Though either the Initial packet
or the Version Negotiation packet that is sent in response could be
lost, the client will send new packets until it successfully receives
a response or it abandons the connection attempt.
6.2.2. Handling Version Negotiation Packets
When the client receives a Version Negotiation packet, it first
checks that the Destination and Source Connection ID fields match the
Source and Destination Connection ID fields in a packet that the
client sent. If this check fails, the packet MUST be discarded.
Once the Version Negotiation packet is determined to be valid, the
client then selects an acceptable protocol version from the list
provided by the server. The client then attempts to create a
connection using that version. Though the contents of the Initial
packet the client sends might not change in response to version
negotiation, a client MUST increase the packet number it uses on
every packet it sends. Packets MUST continue to use long headers and
MUST include the new negotiated protocol version.
The client MUST use the long header format and include its selected
version on all packets until it has 1-RTT keys and it has received a
packet from the server which is not a Version Negotiation packet.
A client MUST NOT change the version it uses unless it is in response
to a Version Negotiation packet from the server. Once a client
receives a packet from the server which is not a Version Negotiation
packet, it MUST discard other Version Negotiation packets on the same
connection. Similarly, a client MUST ignore a Version Negotiation
packet if it has already received and acted on a Version Negotiation
packet.
A client MUST ignore a Version Negotiation packet that lists the
client's chosen version.
Version negotiation packets have no cryptographic protection. The
result of the negotiation MUST be revalidated as part of the
cryptographic handshake (see Section 6.4.4).
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Internet-Draft QUIC Transport Protocol May 20186.2.3. Using Reserved Versions
For a server to use a new version in the future, clients must
correctly handle unsupported versions. To help ensure this, a server
SHOULD include a reserved version (see Section 3) while generating a
Version Negotiation packet.
The design of version negotiation permits a server to avoid
maintaining state for packets that it rejects in this fashion. The
validation of version negotiation (see Section 6.4.4) only validates
the result of version negotiation, which is the same no matter which
reserved version was sent. A server MAY therefore send different
reserved version numbers in the Version Negotiation Packet and in its
transport parameters.
A client MAY send a packet using a reserved version number. This can
be used to solicit a list of supported versions from a server.
6.3. Cryptographic and Transport Handshake
QUIC relies on a combined cryptographic and transport handshake to
minimize connection establishment latency. QUIC allocates stream 0
for the cryptographic handshake. Version 0x00000001 of QUIC uses TLS
1.3 as described in [QUIC-TLS]; a different QUIC version number could
indicate that a different cryptographic handshake protocol is in use.
QUIC provides this stream with reliable, ordered delivery of data.
In return, the cryptographic handshake provides QUIC with:
o authenticated key exchange, where
* a server is always authenticated,
* a client is optionally authenticated,
* every connection produces distinct and unrelated keys,
* keying material is usable for packet protection for both 0-RTT
and 1-RTT packets, and
* 1-RTT keys have forward secrecy
o authenticated values for the transport parameters of the peer (see
Section 6.4)
o authenticated confirmation of version negotiation (see
Section 6.4.4)
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o authenticated negotiation of an application protocol (TLS uses
ALPN [RFC7301] for this purpose)
o for the server, the ability to carry data that provides assurance
that the client can receive packets that are addressed with the
transport address that is claimed by the client (see Section 6.6)
The initial cryptographic handshake message MUST be sent in a single
packet. Any second attempt that is triggered by address validation
MUST also be sent within a single packet. This avoids having to
reassemble a message from multiple packets. Reassembling messages
requires that a server maintain state prior to establishing a
connection, exposing the server to a denial of service risk.
The first client packet of the cryptographic handshake protocol MUST
fit within a 1232 octet QUIC packet payload. This includes overheads
that reduce the space available to the cryptographic handshake
protocol.
Details of how TLS is integrated with QUIC is provided in more detail
in [QUIC-TLS].
6.4. Transport Parameters
During connection establishment, both endpoints make authenticated
declarations of their transport parameters. These declarations are
made unilaterally by each endpoint. Endpoints are required to comply
with the restrictions implied by these parameters; the description of
each parameter includes rules for its handling.
The format of the transport parameters is the TransportParameters
struct from Figure 6. This is described using the presentation
language from Section 3 of [I-D.ietf-tls-tls13].
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completes, the transport parameters declared by the peer are
available. Each endpoint validates the value provided by its peer.
In particular, version negotiation MUST be validated (see
Section 6.4.4) before the connection establishment is considered
properly complete.
Definitions for each of the defined transport parameters are included
in Section 6.4.1. Any given parameter MUST appear at most once in a
given transport parameters extension. An endpoint MUST treat receipt
of duplicate transport parameters as a connection error of type
TRANSPORT_PARAMETER_ERROR.
6.4.1. Transport Parameter Definitions
An endpoint MUST include the following parameters in its encoded
TransportParameters:
initial_max_stream_data (0x0000): The initial stream maximum data
parameter contains the initial value for the maximum data that can
be sent on any newly created stream. This parameter is encoded as
an unsigned 32-bit integer in units of octets. This is equivalent
to an implicit MAX_STREAM_DATA frame (Section 7.7) being sent on
all streams immediately after opening.
initial_max_data (0x0001): The initial maximum data parameter
contains the initial value for the maximum amount of data that can
be sent on the connection. This parameter is encoded as an
unsigned 32-bit integer in units of octets. This is equivalent to
sending a MAX_DATA (Section 7.6) for the connection immediately
after completing the handshake.
idle_timeout (0x0003): The idle timeout is a value in seconds that
is encoded as an unsigned 16-bit integer. The maximum value is
600 seconds (10 minutes).
An endpoint MAY use the following transport parameters:
initial_max_bidi_streams (0x0002): The initial maximum bidirectional
streams parameter contains the initial maximum number of
application-owned bidirectional streams the peer may initiate,
encoded as an unsigned 16-bit integer. If this parameter is
absent or zero, application-owned bidirectional streams cannot be
created until a MAX_STREAM_ID frame is sent. Note that a value of
0 does not prevent the cryptographic handshake stream (that is,
stream 0) from being used. Setting this parameter is equivalent
to sending a MAX_STREAM_ID (Section 7.8) immediately after
completing the handshake containing the corresponding Stream ID.
For example, a value of 0x05 would be equivalent to receiving a
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MAX_STREAM_ID containing 20 when received by a client or 17 when
received by a server.
initial_max_uni_streams (0x0008): The initial maximum unidirectional
streams parameter contains the initial maximum number of
application-owned unidirectional streams the peer may initiate,
encoded as an unsigned 16-bit integer. If this parameter is
absent or zero, unidirectional streams cannot be created until a
MAX_STREAM_ID frame is sent. Setting this parameter is equivalent
to sending a MAX_STREAM_ID (Section 7.8) immediately after
completing the handshake containing the corresponding Stream ID.
For example, a value of 0x05 would be equivalent to receiving a
MAX_STREAM_ID containing 18 when received by a client or 19 when
received by a server.
max_packet_size (0x0005): The maximum packet size parameter places a
limit on the size of packets that the endpoint is willing to
receive, encoded as an unsigned 16-bit integer. This indicates
that packets larger than this limit will be dropped. The default
for this parameter is the maximum permitted UDP payload of 65527.
Values below 1200 are invalid. This limit only applies to
protected packets (Section 4.5).
ack_delay_exponent (0x0007): An 8-bit unsigned integer value
indicating an exponent used to decode the ACK Delay field in the
ACK frame, see Section 7.15. If this value is absent, a default
value of 3 is assumed (indicating a multiplier of 8). The default
value is also used for ACK frames that are sent in Initial,
Handshake, and Retry packets. Values above 20 are invalid.
A server MAY include the following transport parameters:
stateless_reset_token (0x0006): The Stateless Reset Token is used in
verifying a stateless reset, see Section 6.10.4. This parameter
is a sequence of 16 octets.
preferred_address (0x0004): The server's Preferred Address is used
to effect a change in server address at the end of the handshake,
as described in Section 6.9.
A client MUST NOT include a stateless reset token or a preferred
address. A server MUST treat receipt of either transport parameter
as a connection error of type TRANSPORT_PARAMETER_ERROR.
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Internet-Draft QUIC Transport Protocol May 20186.4.2. Values of Transport Parameters for 0-RTT
A client that attempts to send 0-RTT data MUST remember the transport
parameters used by the server. The transport parameters that the
server advertises during connection establishment apply to all
connections that are resumed using the keying material established
during that handshake. Remembered transport parameters apply to the
new connection until the handshake completes and new transport
parameters from the server can be provided.
A server can remember the transport parameters that it advertised, or
store an integrity-protected copy of the values in the ticket and
recover the information when accepting 0-RTT data. A server uses the
transport parameters in determining whether to accept 0-RTT data.
A server MAY accept 0-RTT and subsequently provide different values
for transport parameters for use in the new connection. If 0-RTT
data is accepted by the server, the server MUST NOT reduce any limits
or alter any values that might be violated by the client with its
0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT
set values for initial_max_data or initial_max_stream_data that are
smaller than the remembered value of those parameters. Similarly, a
server MUST NOT reduce the value of initial_max_bidi_streams or
initial_max_uni_streams.
Omitting or setting a zero value for certain transport parameters can
result in 0-RTT data being enabled, but not usable. The following
transport parameters SHOULD be set to non-zero values for 0-RTT:
initial_max_bidi_streams, initial_max_uni_streams, initial_max_data,
initial_max_stream_data.
The value of the server's previous preferred_address MUST NOT be used
when establishing a new connection; rather, the client should wait to
observe the server's new preferred_address value in the handshake.
A server MUST reject 0-RTT data or even abort a handshake if the
implied values for transport parameters cannot be supported.
6.4.3. New Transport Parameters
New transport parameters can be used to negotiate new protocol
behavior. An endpoint MUST ignore transport parameters that it does
not support. Absence of a transport parameter therefore disables any
optional protocol feature that is negotiated using the parameter.
New transport parameters can be registered according to the rules in
Section 13.1.
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Internet-Draft QUIC Transport Protocol May 20186.4.4. Version Negotiation Validation
Though the cryptographic handshake has integrity protection, two
forms of QUIC version downgrade are possible. In the first, an
attacker replaces the QUIC version in the Initial packet. In the
second, a fake Version Negotiation packet is sent by an attacker. To
protect against these attacks, the transport parameters include three
fields that encode version information. These parameters are used to
retroactively authenticate the choice of version (see Section 6.2).
The cryptographic handshake provides integrity protection for the
negotiated version as part of the transport parameters (see
Section 6.4). As a result, attacks on version negotiation by an
attacker can be detected.
The client includes the initial_version field in its transport
parameters. The initial_version is the version that the client
initially attempted to use. If the server did not send a Version
Negotiation packet Section 4.3, this will be identical to the
negotiated_version field in the server transport parameters.
A server that processes all packets in a stateful fashion can
remember how version negotiation was performed and validate the
initial_version value.
A server that does not maintain state for every packet it receives
(i.e., a stateless server) uses a different process. If the
initial_version matches the version of QUIC that is in use, a
stateless server can accept the value.
If the initial_version is different from the version of QUIC that is
in use, a stateless server MUST check that it would have sent a
Version Negotiation packet if it had received a packet with the
indicated initial_version. If a server would have accepted the
version included in the initial_version and the value differs from
the QUIC version that is in use, the server MUST terminate the
connection with a VERSION_NEGOTIATION_ERROR error.
The server includes both the version of QUIC that is in use and a
list of the QUIC versions that the server supports.
The negotiated_version field is the version that is in use. This
MUST be set by the server to the value that is on the Initial packet
that it accepts (not an Initial packet that triggers a Retry or
Version Negotiation packet). A client that receives a
negotiated_version that does not match the version of QUIC that is in
use MUST terminate the connection with a VERSION_NEGOTIATION_ERROR
error code.
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The server includes a list of versions that it would send in any
version negotiation packet (Section 4.3) in the supported_versions
field. The server populates this field even if it did not send a
version negotiation packet.
The client validates that the negotiated_version is included in the
supported_versions list and - if version negotiation was performed -
that it would have selected the negotiated version. A client MUST
terminate the connection with a VERSION_NEGOTIATION_ERROR error code
if the current QUIC version is not listed in the supported_versions
list. A client MUST terminate with a VERSION_NEGOTIATION_ERROR error
code if version negotiation occurred but it would have selected a
different version based on the value of the supported_versions list.
When an endpoint accepts multiple QUIC versions, it can potentially
interpret transport parameters as they are defined by any of the QUIC
versions it supports. The version field in the QUIC packet header is
authenticated using transport parameters. The position and the
format of the version fields in transport parameters MUST either be
identical across different QUIC versions, or be unambiguously
different to ensure no confusion about their interpretation. One way
that a new format could be introduced is to define a TLS extension
with a different codepoint.
6.5. Stateless Retries
A server can process an initial cryptographic handshake messages from
a client without committing any state. This allows a server to
perform address validation (Section 6.6), or to defer connection
establishment costs.
A server that generates a response to an initial packet without
retaining connection state MUST use the Retry packet (Section 4.4.2).
This packet causes a client to reset its transport state and to
continue the connection attempt with new connection state while
maintaining the state of the cryptographic handshake.
A server MUST NOT send multiple Retry packets in response to a client
handshake packet. Thus, any cryptographic handshake message that is
sent MUST fit within a single packet.
In TLS, the Retry packet type is used to carry the HelloRetryRequest
message.
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Internet-Draft QUIC Transport Protocol May 20186.6. Proof of Source Address Ownership
Transport protocols commonly spend a round trip checking that a
client owns the transport address (IP and port) that it claims.
Verifying that a client can receive packets sent to its claimed
transport address protects against spoofing of this information by
malicious clients.
This technique is used primarily to avoid QUIC from being used for
traffic amplification attack. In such an attack, a packet is sent to
a server with spoofed source address information that identifies a
victim. If a server generates more or larger packets in response to
that packet, the attacker can use the server to send more data toward
the victim than it would be able to send on its own.
Several methods are used in QUIC to mitigate this attack. Firstly,
the initial handshake packet is padded to at least 1200 octets. This
allows a server to send a similar amount of data without risking
causing an amplification attack toward an unproven remote address.
A server eventually confirms that a client has received its messages
when the cryptographic handshake successfully completes. This might
be insufficient, either because the server wishes to avoid the
computational cost of completing the handshake, or it might be that
the size of the packets that are sent during the handshake is too
large. This is especially important for 0-RTT, where the server
might wish to provide application data traffic - such as a response
to a request - in response to the data carried in the early data from
the client.
To send additional data prior to completing the cryptographic
handshake, the server then needs to validate that the client owns the
address that it claims.
Source address validation is therefore performed during the
establishment of a connection. TLS provides the tools that support
the feature, but basic validation is performed by the core transport
protocol.
A different type of source address validation is performed after a
connection migration, see Section 6.7.
6.6.1. Client Address Validation Procedure
QUIC uses token-based address validation. Any time the server wishes
to validate a client address, it provides the client with a token.
As long as the token cannot be easily guessed (see Section 6.6.3), if
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the client is able to return that token, it proves to the server that
it received the token.
During the processing of the cryptographic handshake messages from a
client, TLS will request that QUIC make a decision about whether to
proceed based on the information it has. TLS will provide QUIC with
any token that was provided by the client. For an initial packet,
QUIC can decide to abort the connection, allow it to proceed, or
request address validation.
If QUIC decides to request address validation, it provides the
cryptographic handshake with a token. The contents of this token are
consumed by the server that generates the token, so there is no need
for a single well-defined format. A token could include information
about the claimed client address (IP and port), a timestamp, and any
other supplementary information the server will need to validate the
token in the future.
The cryptographic handshake is responsible for enacting validation by
sending the address validation token to the client. A legitimate
client will include a copy of the token when it attempts to continue
the handshake. The cryptographic handshake extracts the token then
asks QUIC a second time whether the token is acceptable. In
response, QUIC can either abort the connection or permit it to
proceed.
A connection MAY be accepted without address validation - or with
only limited validation - but a server SHOULD limit the data it sends
toward an unvalidated address. Successful completion of the
cryptographic handshake implicitly provides proof that the client has
received packets from the server.
6.6.2. Address Validation on Session Resumption
A server MAY provide clients with an address validation token during
one connection that can be used on a subsequent connection. Address
validation is especially important with 0-RTT because a server
potentially sends a significant amount of data to a client in
response to 0-RTT data.
A different type of token is needed when resuming. Unlike the token
that is created during a handshake, there might be some time between
when the token is created and when the token is subsequently used.
Thus, a resumption token SHOULD include an expiration time. It is
also unlikely that the client port number is the same on two
different connections; validating the port is therefore unlikely to
be successful.
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This token can be provided to the cryptographic handshake immediately
after establishing a connection. QUIC might also generate an updated
token if significant time passes or the client address changes for
any reason (see Section 6.8). The cryptographic handshake is
responsible for providing the client with the token. In TLS the
token is included in the ticket that is used for resumption and
0-RTT, which is carried in a NewSessionTicket message.
6.6.3. Address Validation Token Integrity
An address validation token MUST be difficult to guess. Including a
large enough random value in the token would be sufficient, but this
depends on the server remembering the value it sends to clients.
A token-based scheme allows the server to offload any state
associated with validation to the client. For this design to work,
the token MUST be covered by integrity protection against
modification or falsification by clients. Without integrity
protection, malicious clients could generate or guess values for
tokens that would be accepted by the server. Only the server
requires access to the integrity protection key for tokens.
In TLS the address validation token is often bundled with the
information that TLS requires, such as the resumption secret. In
this case, adding integrity protection can be delegated to the
cryptographic handshake protocol, avoiding redundant protection. If
integrity protection is delegated to the cryptographic handshake, an
integrity failure will result in immediate cryptographic handshake
failure. If integrity protection is performed by QUIC, QUIC MUST
abort the connection if the integrity check fails with a
PROTOCOL_VIOLATION error code.
6.7. Path Validation
Path validation is used by an endpoint to verify reachability of a
peer over a specific path. That is, it tests reachability between a
specific local address and a specific peer address, where an address
is the two-tuple of IP address and port. Path validation tests that
packets can be both sent to and received from a peer.
Path validation is used during connection migration (see Section 6.8
and Section 6.9) by the migrating endpoint to verify reachability of
a peer from a new local address. Path validation is also used by the
peer to verify that the migrating endpoint is able to receive packets
sent to the its new address. That is, that the packets received from
the migrating endpoint do not carry a spoofed source address.
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Path validation can be used at any time by either endpoint. For
instance, an endpoint might check that a peer is still in possession
of its address after a period of quiescence.
Path validation is not designed as a NAT traversal mechanism. Though
the mechanism described here might be effective for the creation of
NAT bindings that support NAT traversal, the expectation is that one
or other peer is able to receive packets without first having sent a
packet on that path. Effective NAT traversal needs additional
synchronization mechanisms that are not provided here.
An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that
are used for path validation with other frames. For instance, an
endpoint may pad a packet carrying a PATH_CHALLENGE for PMTU
discovery, or an endpoint may bundle a PATH_RESPONSE with its own
PATH_CHALLENGE.
6.7.1. Initiation
To initiate path validation, an endpoint sends a PATH_CHALLENGE frame
containing a random payload on the path to be validated.
An endpoint MAY send additional PATH_CHALLENGE frames to handle
packet loss. An endpoint SHOULD NOT send a PATH_CHALLENGE more
frequently than it would an Initial packet, ensuring that connection
migration is no more load on a new path than establishing a new
connection.
The endpoint MUST use fresh random data in every PATH_CHALLENGE frame
so that it can associate the peer's response with the causative
PATH_CHALLENGE.
6.7.2. Response
On receiving a PATH_CHALLENGE frame, an endpoint MUST respond
immediately by echoing the data contained in the PATH_CHALLENGE frame
in a PATH_RESPONSE frame, with the following stipulation. Since a
PATH_CHALLENGE might be sent from a spoofed address, an endpoint MAY
limit the rate at which it sends PATH_RESPONSE frames and MAY
silently discard PATH_CHALLENGE frames that would cause it to respond
at a higher rate.
To ensure that packets can be both sent to and received from the
peer, the PATH_RESPONSE MUST be sent on the same path as the
triggering PATH_CHALLENGE: from the same local address on which the
PATH_CHALLENGE was received, to the same remote address from which
the PATH_CHALLENGE was received.
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A new address is considered valid when a PATH_RESPONSE frame is
received containing data that was sent in a previous PATH_CHALLENGE.
Receipt of an acknowledgment for a packet containing a PATH_CHALLENGE
frame is not adequate validation, since the acknowledgment can be
spoofed by a malicious peer.
For path validation to be successful, a PATH_RESPONSE frame MUST be
received from the same remote address to which the corresponding
PATH_CHALLENGE was sent. If a PATH_RESPONSE frame is received from a
different remote address than the one to which the PATH_CHALLENGE was
sent, path validation is considered to have failed, even if the data
matches that sent in the PATH_CHALLENGE.
Additionally, the PATH_RESPONSE frame MUST be received on the same
local address from which the corresponding PATH_CHALLENGE was sent.
If a PATH_RESPONSE frame is received on a different local address
than the one from which the PATH_CHALLENGE was sent, path validation
is considered to have failed, even if the data matches that sent in
the PATH_CHALLENGE. Thus, the endpoint considers the path to be
valid when a PATH_RESPONSE frame is received on the same path with
the same payload as the PATH_CHALLENGE frame.
6.7.4. Abandonment
An endpoint SHOULD abandon path validation after sending some number
of PATH_CHALLENGE frames or after some time has passed. When setting
this timer, implementations are cautioned that the new path could
have a longer round-trip time than the original.
Note that the endpoint might receive packets containing other frames
on the new path, but a PATH_RESPONSE frame with appropriate data is
required for path validation to succeed.
If path validation fails, the path is deemed unusable. This does not
necessarily imply a failure of the connection - endpoints can
continue sending packets over other paths as appropriate. If no
paths are available, an endpoint can wait for a new path to become
available or close the connection.
A path validation might be abandoned for other reasons besides
failure. Primarily, this happens if a connection migration to a new
path is initiated while a path validation on the old path is in
progress.
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Internet-Draft QUIC Transport Protocol May 20186.8. Connection Migration
QUIC allows connections to survive changes to endpoint addresses
(that is, IP address and/or port), such as those caused by a endpoint
migrating to a new network. This section describes the process by
which an endpoint migrates to a new address.
An endpoint MUST NOT initiate connection migration before the
handshake is finished and the endpoint has 1-RTT keys.
This document limits migration of connections to new client
addresses, except as described in Section 6.9. Clients are
responsible for initiating all migrations. Servers do not send non-
probing packets (see Section 6.8.1) toward a client address until it
sees a non-probing packet from that address. If a client receives
packets from an unknown server address, the client MAY discard these
packets.
6.8.1. Probing a New Path
An endpoint MAY probe for peer reachability from a new local address
using path validation Section 6.7 prior to migrating the connection
to the new local address. Failure of path validation simply means
that the new path is not usable for this connection. Failure to
validate a path does not cause the connection to end unless there are
no valid alternative paths available.
An endpoint uses a new connection ID for probes sent from a new local
address, see Section 6.8.5 for further discussion.
Receiving a PATH_CHALLENGE frame from a peer indicates that the peer
is probing for reachability on a path. An endpoint sends a
PATH_RESPONSE in response as per Section 6.7.
PATH_CHALLENGE, PATH_RESPONSE, and PADDING frames are "probing
frames", and all other frames are "non-probing frames". A packet
containing only probing frames is a "probing packet", and a packet
containing any other frame is a "non-probing packet".
6.8.2. Initiating Connection Migration
A endpoint can migrate a connection to a new local address by sending
packets containing frames other than probing frames from that
address.
Each endpoint validates its peer's address during connection
establishment. Therefore, a migrating endpoint can send to its peer
knowing that the peer is willing to receive at the peer's current
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address. Thus an endpoint can migrate to a new local address without
first validating the peer's address.
When migrating, the new path might not support the endpoint's current
sending rate. Therefore, the endpoint resets its congestion
controller, as described in Section 6.8.4.
Receiving acknowledgments for data sent on the new path serves as
proof of the peer's reachability from the new address. Note that
since acknowledgments may be received on any path, return
reachability on the new path is not established. To establish return
reachability on the new path, an endpoint MAY concurrently initiate
path validation Section 6.7 on the new path.
6.8.3. Responding to Connection Migration
Receiving a packet from a new peer address containing a non-probing
frame indicates that the peer has migrated to that address.
In response to such a packet, an endpoint MUST start sending
subsequent packets to the new peer address and MUST initiate path
validation (Section 6.7) to verify the peer's ownership of the
unvalidated address.
An endpoint MAY send data to an unvalidated peer address, but it MUST
protect against potential attacks as described in Section 6.8.3.1 and
Section 6.8.3.2. An endpoint MAY skip validation of a peer address
if that address has been seen recently.
An endpoint only changes the address that it sends packets to in
response to the highest-numbered non-probing packet. This ensures
that an endpoint does not send packets to an old peer address in the
case that it receives reordered packets.
After changing the address to which it sends non-probing packets, an
endpoint could abandon any path validation for other addresses.
Receiving a packet from a new peer address might be the result of a
NAT rebinding at the peer.
After verifying a new client address, the server SHOULD send new
address validation tokens (Section 6.6) to the client.
6.8.3.1. Handling Address Spoofing by a Peer
It is possible that a peer is spoofing its source address to cause an
endpoint to send excessive amounts of data to an unwilling host. If
the endpoint sends significantly more data than the spoofing peer,
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connection migration might be used to amplify the volume of data that
an attacker can generate toward a victim.
As described in Section 6.8.3, an endpoint is required to validate a
peer's new address to confirm the peer's possession of the new
address. Until a peer's address is deemed valid, an endpoint MUST
limit the rate at which it sends data to this address. The endpoint
MUST NOT send more than a minimum congestion window's worth of data
per estimated round-trip time (kMinimumWindow, as defined in
[QUIC-RECOVERY]). In the absence of this limit, an endpoint risks
being used for a denial of service attack against an unsuspecting
victim. Note that since the endpoint will not have any round-trip
time measurements to this address, the estimate SHOULD be the default
initial value (see [QUIC-RECOVERY]).
If an endpoint skips validation of a peer address as described in
Section 6.8.3, it does not need to limit its sending rate.
6.8.3.2. Handling Address Spoofing by an On-path Attacker
An on-path attacker could cause a spurious connection migration by
copying and forwarding a packet with a spoofed address such that it
arrives before the original packet. The packet with the spoofed
address will be seen to come from a migrating connection, and the
original packet will be seen as a duplicate and dropped. After a
spurious migration, validation of the source address will fail
because the entity at the source address does not have the necessary
cryptographic keys to read or respond to the PATH_CHALLENGE frame
that is sent to it even if it wanted to.
To protect the connection from failing due to such a spurious
migration, an endpoint MUST revert to using the last validated peer
address when validation of a new peer address fails.
If an endpoint has no state about the last validated peer address, it
MUST close the connection silently by discarding all connection
state. This results in new packets on the connection being handled
generically. For instance, an endpoint MAY send a stateless reset in
response to any further incoming packets.
Note that receipt of packets with higher packet numbers from the
legitimate peer address will trigger another connection migration.
This will cause the validation of the address of the spurious
migration to be abandoned.
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The capacity available on the new path might not be the same as the
old path. Packets sent on the old path SHOULD NOT contribute to
congestion control or RTT estimation for the new path.
On confirming a peer's ownership of its new address, an endpoint
SHOULD immediately reset the congestion controller and round-trip
time estimator for the new path.
An endpoint MUST NOT return to the send rate used for the previous
path unless it is reasonably sure that the previous send rate is
valid for the new path. For instance, a change in the client's port
number is likely indicative of a rebinding in a middlebox and not a
complete change in path. This determination likely depends on
heuristics, which could be imperfect; if the new path capacity is
significantly reduced, ultimately this relies on the congestion
controller responding to congestion signals and reducing send rates
appropriately.
There may be apparent reordering at the receiver when an endpoint
sends data and probes from/to multiple addresses during the migration
period, since the two resulting paths may have different round-trip
times. A receiver of packets on multiple paths will still send ACK
frames covering all received packets.
While multiple paths might be used during connection migration, a
single congestion control context and a single loss recovery context
(as described in [QUIC-RECOVERY]) may be adequate. A sender can make
exceptions for probe packets so that their loss detection is
independent and does not unduly cause the congestion controller to
reduce its sending rate. An endpoint might arm a separate alarm when
a PATH_CHALLENGE is sent, which is disarmed when the corresponding
PATH_RESPONSE is received. If the alarm fires before the
PATH_RESPONSE is received, the endpoint might send a new
PATH_CHALLENGE, and restart the alarm for a longer period of time.
6.8.5. Privacy Implications of Connection Migration
Using a stable connection ID on multiple network paths allows a
passive observer to correlate activity between those paths. An
endpoint that moves between networks might not wish to have their
activity correlated by any entity other than their peer. The
NEW_CONNECTION_ID message can be sent to provide an unlinkable
connection ID for use in case a peer wishes to explicitly break
linkability between two points of network attachment.
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An endpoint that does not require the use of a connection ID should
not request that its peer use a connection ID. Such an endpoint does
not need to provide new connection IDs using the NEW_CONNECTION_ID
frame.
An endpoint might need to send packets on multiple networks without
receiving any response from its peer. To ensure that the endpoint is
not linkable across each of these changes, a new connection ID is
needed for each network. To support this, multiple NEW_CONNECTION_ID
messages are needed. Each NEW_CONNECTION_ID is marked with a
sequence number. Connection IDs MUST be used in the order in which
they are numbered.
An endpoint that to break linkability upon changing networks MUST use
a previously unused connection ID provided by its peer. Protection
of packet numbers ensures that packet numbers cannot be used to
correlate connections. Other properties of packets, such as timing
and size, might be used to correlate activity, but no explicit
correlation can be used to link activity across paths.
Clients MAY change connection ID at any time based on implementation-
specific concerns. For example, after a period of network inactivity
NAT rebinding might occur when the client begins sending data again.
A client might wish to reduce linkability by employing a new
connection ID and source UDP port when sending traffic after a period
of inactivity. Changing the UDP port from which it sends packets at
the same time might cause the packet to appear as a connection
migration. This ensures that the mechanisms that support migration
are exercised even for clients that don't experience NAT rebindings
or genuine migrations. Changing port number can cause a peer to
reset its congestion state (see Section 6.8.4), so the port SHOULD
only be changed infrequently.
An endpoint that receives a successfully authenticated packet with a
previously unused connection ID MUST use the next available
connection ID for any packets it sends to that address. To avoid
changing connection IDs multiple times when packets arrive out of
order, endpoints MUST change only in response to a packet that
increases the largest received packet number. Failing to do this
could allow for use of that connection ID to link activity on new
paths. There is no need to move to a new connection ID if the
address of a peer changes without also changing the connection ID.
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Internet-Draft QUIC Transport Protocol May 20186.9. Server's Preferred Address
QUIC allows servers to accept connections on one IP address and
attempt to transfer these connections to a more preferred address
shortly after the handshake. This is particularly useful when
clients initially connect to an address shared by multiple servers
but would prefer to use a unicast address to ensure connection
stability. This section describes the protocol for migrating a
connection to a preferred server address.
Migrating a connection to a new server address mid-connection is left
for future work. If a client receives packets from a new server
address not indicated by the preferred_address transport parameter,
the client SHOULD discard these packets.
6.9.1. Communicating A Preferred Address
A server conveys a preferred address by including the
preferred_address transport parameter in the TLS handshake.
Once the handshake is finished, the client SHOULD initiate path
validation (see Section 6.7) of the server's preferred address using
the connection ID provided in the preferred_address transport
parameter.
If path validation succeeds, the client SHOULD immediately begin
sending all future packets to the new server address using the new
connection ID and discontinue use of the old server address. If path
validation fails, the client MUST continue sending all future packets
to the server's original IP address.
6.9.2. Responding to Connection Migration
A server might receive a packet addressed to its preferred IP address
at any time after the handshake is completed. If this packet
contains a PATH_CHALLENGE frame, the server sends a PATH_RESPONSE
frame as per Section 6.7, but the server MUST continue sending all
other packets from its original IP address.
The server SHOULD also initiate path validation of the client using
its preferred address and the address from which it received the
client probe. This helps to guard against spurious migration
initiated by an attacker.
Once the server has completed its path validation and has received a
non-probing packet with a new largest packet number on its preferred
address, the server begins sending to the client exclusively from its
preferred IP address. It SHOULD drop packets for this connection
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received on the old IP address, but MAY continue to process delayed
packets.
6.9.3. Interaction of Client Migration and Preferred Address
A client might need to perform a connection migration before it has
migrated to the server's preferred address. In this case, the client
SHOULD perform path validation to both the original and preferred
server address from the client's new address concurrently.
If path validation of the server's preferred address succeeds, the
client MUST abandon validation of the original address and migrate to
using the server's preferred address. If path validation of the
server's preferred address fails, but validation of the server's
original address succeeds, the client MAY migrate to using the
original address from the client's new address.
If the connection to the server's preferred address is not from the
same client address, the server MUST protect against potential
attacks as described in Section 6.8.3.1 and Section 6.8.3.2. In
addition to intentional simultaneous migration, this might also occur
because the client's access network used a different NAT binding for
the server's preferred address.
Servers SHOULD initiate path validation to the client's new address
upon receiving a probe packet from a different address. Servers MUST
NOT send more than a minimum congestion window's worth of non-probing
packets to the new address before path validation is complete.
6.10. Connection Termination
Connections should remain open until they become idle for a pre-
negotiated period of time. A QUIC connection, once established, can
be terminated in one of three ways:
o idle timeout (Section 6.10.2)
o immediate close (Section 6.10.3)
o stateless reset (Section 6.10.4)
6.10.1. Closing and Draining Connection States
The closing and draining connection states exist to ensure that
connections close cleanly and that delayed or reordered packets are
properly discarded. These states SHOULD persist for three times the
current Retransmission Timeout (RTO) interval as defined in
[QUIC-RECOVERY].
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An endpoint enters a closing period after initiating an immediate
close (Section 6.10.3). While closing, an endpoint MUST NOT send
packets unless they contain a CONNECTION_CLOSE or APPLICATION_CLOSE
frame (see Section 6.10.3 for details).
In the closing state, only a packet containing a closing frame can be
sent. An endpoint retains only enough information to generate a
packet containing a closing frame and to identify packets as
belonging to the connection. The connection ID and QUIC version is
sufficient information to identify packets for a closing connection;
an endpoint can discard all other connection state. An endpoint MAY
retain packet protection keys for incoming packets to allow it to
read and process a closing frame.
The draining state is entered once an endpoint receives a signal that
its peer is closing or draining. While otherwise identical to the
closing state, an endpoint in the draining state MUST NOT send any
packets. Retaining packet protection keys is unnecessary once a
connection is in the draining state.
An endpoint MAY transition from the closing period to the draining
period if it can confirm that its peer is also closing or draining.
Receiving a closing frame is sufficient confirmation, as is receiving
a stateless reset. The draining period SHOULD end when the closing
period would have ended. In other words, the endpoint can use the
same end time, but cease retransmission of the closing packet.
Disposing of connection state prior to the end of the closing or
draining period could cause delayed or reordered packets to be
handled poorly. Endpoints that have some alternative means to ensure
that late-arriving packets on the connection do not create QUIC
state, such as those that are able to close the UDP socket, MAY use
an abbreviated draining period which can allow for faster resource
recovery. Servers that retain an open socket for accepting new
connections SHOULD NOT exit the closing or draining period early.
Once the closing or draining period has ended, an endpoint SHOULD
discard all connection state. This results in new packets on the
connection being handled generically. For instance, an endpoint MAY
send a stateless reset in response to any further incoming packets.
The draining and closing periods do not apply when a stateless reset
(Section 6.10.4) is sent.
An endpoint is not expected to handle key updates when it is closing
or draining. A key update might prevent the endpoint from moving
from the closing state to draining, but it otherwise has no impact.
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An endpoint could receive packets from a new source address,
indicating a client connection migration (Section 6.8), while in the
closing period. An endpoint in the closing state MUST strictly limit
the number of packets it sends to this new address until the address
is validated (see Section 6.7). A server in the closing state MAY
instead choose to discard packets received from a new source address.
6.10.2. Idle Timeout
A connection that remains idle for longer than the idle timeout (see
Section 6.4.1) is closed. A connection enters the draining state
when the idle timeout expires.
The time at which an idle timeout takes effect won't be perfectly
synchronized on both endpoints. An endpoint that sends packets near
the end of an idle period could have those packets discarded if its
peer enters the draining state before the packet is received.
6.10.3. Immediate Close
An endpoint sends a closing frame, either CONNECTION_CLOSE or
APPLICATION_CLOSE, to terminate the connection immediately. Either
closing frame causes all streams to immediately become closed; open
streams can be assumed to be implicitly reset.
After sending a closing frame, endpoints immediately enter the
closing state. During the closing period, an endpoint that sends a
closing frame SHOULD respond to any packet that it receives with
another packet containing a closing frame. To minimize the state
that an endpoint maintains for a closing connection, endpoints MAY
send the exact same packet. However, endpoints SHOULD limit the
number of packets they generate containing a closing frame. For
instance, an endpoint could progressively increase the number of
packets that it receives before sending additional packets or
increase the time between packets.
Note: Allowing retransmission of a packet contradicts other advice
in this document that recommends the creation of new packet
numbers for every packet. Sending new packet numbers is primarily
of advantage to loss recovery and congestion control, which are
not expected to be relevant for a closed connection.
Retransmitting the final packet requires less state.
After receiving a closing frame, endpoints enter the draining state.
An endpoint that receives a closing frame MAY send a single packet
containing a closing frame before entering the draining state, using
a CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An
endpoint MUST NOT send further packets, which could result in a
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constant exchange of closing frames until the closing period on
either peer ended.
An immediate close can be used after an application protocol has
arranged to close a connection. This might be after the application
protocols negotiates a graceful shutdown. The application protocol
exchanges whatever messages that are needed to cause both endpoints
to agree to close the connection, after which the application
requests that the connection be closed. The application protocol can
use an APPLICATION_CLOSE message with an appropriate error code to
signal closure.
6.10.4. Stateless Reset
A stateless reset is provided as an option of last resort for a
server that does not have access to the state of a connection. A
server crash or outage might result in clients continuing to send
data to a server that is unable to properly continue the connection.
A server that wishes to communicate a fatal connection error MUST use
a closing frame if it has sufficient state to do so.
To support this process, the server sends a stateless_reset_token
value during the handshake in the transport parameters. This value
is protected by encryption, so only client and server know this
value.
A server that receives packets that it cannot process sends a packet
in the following layout:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|0|K| Type (6) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Random Octets (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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This design ensures that a stateless reset packet is - to the extent
possible - indistinguishable from a regular packet with a short
header.
A server generates a random 18-octet Destination Connection ID field.
For a client that depends on the server including a connection ID,
this will mean that this value differs from previous packets. Ths
results in two problems:
o The packet might not reach the client. If the Destination
Connection ID is critical for routing toward the client, then this
packet could be incorrectly routed. This causes the stateless
reset to be ineffective in causing errors to be quickly detected
and recovered. In this case, clients will need to rely on other
methods - such as timers - to detect that the connection has
failed.
o The randomly generated connection ID can be used by entities other
than the client to identify this as a potential stateless reset.
A server that occasionally uses different connection IDs might
introduce some uncertainty about this.
The Packet Number field is set to a randomized value. The server
SHOULD send a packet with a short header and a packet number length
of 1 octet. Using the shortest possible packet number encoding
minimizes the perceived gap between the last packet that the server
sent and this packet. A server MAY indicate a different packet
number length, but a longer packet number encoding might allow this
message to be identified as a stateless reset more easily using
heuristics.
After the Packet Number, the server pads the message with an
arbitrary number of octets containing random values.
Finally, the last 16 octets of the packet are set to the value of the
Stateless Reset Token.
A stateless reset is not appropriate for signaling error conditions.
An endpoint that wishes to communicate a fatal connection error MUST
use a CONNECTION_CLOSE or APPLICATION_CLOSE frame if it has
sufficient state to do so.
This stateless reset design is specific to QUIC version 1. A server
that supports multiple versions of QUIC needs to generate a stateless
reset that will be accepted by clients that support any version that
the server might support (or might have supported prior to losing
state). Designers of new versions of QUIC need to be aware of this
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and either reuse this design, or use a portion of the packet other
than the last 16 octets for carrying data.
6.10.4.1. Detecting a Stateless Reset
A client detects a potential stateless reset when a packet with a
short header either cannot be decrypted or is marked as a duplicate
packet. The client then compares the last 16 octets of the packet
with the Stateless Reset Token provided by the server in its
transport parameters. If these values are identical, the client MUST
enter the draining period and not send any further packets on this
connection. If the comparison fails, the packet can be discarded.
6.10.4.2. Calculating a Stateless Reset Token
The stateless reset token MUST be difficult to guess. In order to
create a Stateless Reset Token, a server could randomly generate
[RFC4086] a secret for every connection that it creates. However,
this presents a coordination problem when there are multiple servers
in a cluster or a storage problem for a server that might lose state.
Stateless reset specifically exists to handle the case where state is
lost, so this approach is suboptimal.
A single static key can be used across all connections to the same
endpoint by generating the proof using a second iteration of a
preimage-resistant function that takes three inputs: the static key,
the server's connection ID (see Section 4.7), and an identifier for
the server instance. A server could use HMAC [RFC2104] (for example,
HMAC(static_key, server_id || connection_id)) or HKDF [RFC5869] (for
example, using the static key as input keying material, with server
and connection identifiers as salt). The output of this function is
truncated to 16 octets to produce the Stateless Reset Token for that
connection.
A server that loses state can use the same method to generate a valid
Stateless Reset Secret. The connection ID comes from the packet that
the server receives.
This design relies on the client always sending a connection ID in
its packets so that the server can use the connection ID from a
packet to reset the connection. A server that uses this design
cannot allow clients to use a zero-length connection ID.
Revealing the Stateless Reset Token allows any entity to terminate
the connection, so a value can only be used once. This method for
choosing the Stateless Reset Token means that the combination of
server instance, connection ID, and static key cannot occur for
another connection. A connection ID from a connection that is reset
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Error codes (Section 11.3) are described using integers, but do not
use this encoding.
7.2. PADDING Frame
The PADDING frame (type=0x00) has no semantic value. PADDING frames
can be used to increase the size of a packet. Padding can be used to
increase an initial client packet to the minimum required size, or to
provide protection against traffic analysis for protected packets.
A PADDING frame has no content. That is, a PADDING frame consists of
the single octet that identifies the frame as a PADDING frame.
7.3. RST_STREAM Frame
An endpoint may use a RST_STREAM frame (type=0x01) to abruptly
terminate a stream.
After sending a RST_STREAM, an endpoint ceases transmission and
retransmission of STREAM frames on the identified stream. A receiver
of RST_STREAM can discard any data that it already received on that
stream.
An endpoint that receives a RST_STREAM frame for a send-only stream
MUST terminate the connection with error PROTOCOL_VIOLATION.
The RST_STREAM frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Final Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Stream ID: A variable-length integer encoding of the Stream ID of
the stream being terminated.
Application Protocol Error Code: A 16-bit application protocol error
code (see Section 11.4) which indicates why the stream is being
closed.
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Final Offset: A variable-length integer indicating the absolute byte
offset of the end of data written on this stream by the RST_STREAM
sender.
7.4. CONNECTION_CLOSE frame
An endpoint sends a CONNECTION_CLOSE frame (type=0x02) to notify its
peer that the connection is being closed. CONNECTION_CLOSE is used
to signal errors at the QUIC layer, or the absence of errors (with
the NO_ERROR code).
If there are open streams that haven't been explicitly closed, they
are implicitly closed when the connection is closed.
The CONNECTION_CLOSE frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (16) | Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields of a CONNECTION_CLOSE frame are as follows:
Error Code: A 16-bit error code which indicates the reason for
closing this connection. CONNECTION_CLOSE uses codes from the
space defined in Section 11.3 (APPLICATION_CLOSE uses codes from
the application protocol error code space, see Section 11.4).
Reason Phrase Length: A variable-length integer specifying the
length of the reason phrase in bytes. Note that a
CONNECTION_CLOSE frame cannot be split between packets, so in
practice any limits on packet size will also limit the space
available for a reason phrase.
Reason Phrase: A human-readable explanation for why the connection
was closed. This can be zero length if the sender chooses to not
give details beyond the Error Code. This SHOULD be a UTF-8
encoded string [RFC3629].
7.5. APPLICATION_CLOSE frame
An APPLICATION_CLOSE frame (type=0x03) uses the same format as the
CONNECTION_CLOSE frame (Section 7.4), except that it uses error codes
from the application protocol error code space (Section 11.4) instead
of the transport error code space.
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Other than the error code space, the format and semantics of the
APPLICATION_CLOSE frame are identical to the CONNECTION_CLOSE frame.
7.6. MAX_DATA Frame
The MAX_DATA frame (type=0x04) is used in flow control to inform the
peer of the maximum amount of data that can be sent on the connection
as a whole.
The frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_DATA frame are as follows:
Maximum Data: A variable-length integer indicating the maximum
amount of data that can be sent on the entire connection, in units
of octets.
All data sent in STREAM frames counts toward this limit, with the
exception of data on stream 0. The sum of the largest received
offsets on all streams - including streams in terminal states, but
excluding stream 0 - MUST NOT exceed the value advertised by a
receiver. An endpoint MUST terminate a connection with a
QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA error if it receives more
data than the maximum data value that it has sent, unless this is a
result of a change in the initial limits (see Section 6.4.2).
7.7. MAX_STREAM_DATA Frame
The MAX_STREAM_DATA frame (type=0x05) is used in flow control to
inform a peer of the maximum amount of data that can be sent on a
stream.
An endpoint that receives a MAX_STREAM_DATA frame for a receive-only
stream MUST terminate the connection with error PROTOCOL_VIOLATION.
An endpoint that receives a MAX_STREAM_DATA frame for a send-only
stream it has not opened MUST terminate the connection with error
PROTOCOL_VIOLATION.
Note that an endpoint may legally receive a MAX_STREAM_DATA frame on
a bidirectional stream it has not opened.
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The frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_DATA frame are as follows:
Stream ID: The stream ID of the stream that is affected encoded as a
variable-length integer.
Maximum Stream Data: A variable-length integer indicating the
maximum amount of data that can be sent on the identified stream,
in units of octets.
When counting data toward this limit, an endpoint accounts for the
largest received offset of data that is sent or received on the
stream. Loss or reordering can mean that the largest received offset
on a stream can be greater than the total size of data received on
that stream. Receiving STREAM frames might not increase the largest
received offset.
The data sent on a stream MUST NOT exceed the largest maximum stream
data value advertised by the receiver. An endpoint MUST terminate a
connection with a FLOW_CONTROL_ERROR error if it receives more data
than the largest maximum stream data that it has sent for the
affected stream, unless this is a result of a change in the initial
limits (see Section 6.4.2).
7.8. MAX_STREAM_ID Frame
The MAX_STREAM_ID frame (type=0x06) informs the peer of the maximum
stream ID that they are permitted to open.
The frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_ID frame are as follows:
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Maximum Stream ID: ID of the maximum unidirectional or bidirectional
peer-initiated stream ID for the connection encoded as a variable-
length integer. The limit applies to unidirectional steams if the
second least signification bit of the stream ID is 1, and applies
to bidirectional streams if it is 0.
Loss or reordering can mean that a MAX_STREAM_ID frame can be
received which states a lower stream limit than the client has
previously received. MAX_STREAM_ID frames which do not increase the
maximum stream ID MUST be ignored.
A peer MUST NOT initiate a stream with a higher stream ID than the
greatest maximum stream ID it has received. An endpoint MUST
terminate a connection with a STREAM_ID_ERROR error if a peer
initiates a stream with a higher stream ID than it has sent, unless
this is a result of a change in the initial limits (see
Section 6.4.2).
7.9. PING Frame
Endpoints can use PING frames (type=0x07) to verify that their peers
are still alive or to check reachability to the peer. The PING frame
contains no additional fields.
The receiver of a PING frame simply needs to acknowledge the packet
containing this frame.
The PING frame can be used to keep a connection alive when an
application or application protocol wishes to prevent the connection
from timing out. An application protocol SHOULD provide guidance
about the conditions under which generating a PING is recommended.
This guidance SHOULD indicate whether it is the client or the server
that is expected to send the PING. Having both endpoints send PING
frames without coordination can produce an excessive number of
packets and poor performance.
A connection will time out if no packets are sent or received for a
period longer than the time specified in the idle_timeout transport
parameter (see Section 6.10). However, state in middleboxes might
time out earlier than that. Though REQ-5 in [RFC4787] recommends a 2
minute timeout interval, experience shows that sending packets every
15 to 30 seconds is necessary to prevent the majority of middleboxes
from losing state for UDP flows.
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Sequence: A variable-length integer. This value starts at 0 and
increases by 1 for each connection ID that is provided by the
server. The connection ID that is assigned during the handshake
is assumed to have a sequence of -1. That is, the value selected
during the handshake comes immediately before the first value that
a server can send.
Length: An 8-bit unsigned integer containing the length of the
connection ID. Values less than 4 and greater than 18 are invalid
and MUST be treated as a connection error of type
PROTOCOL_VIOLATION.
Connection ID: A connection ID of the specified length.
Stateless Reset Token: A 128-bit value that will be used to for a
stateless reset when the associated connection ID is used (see
Section 6.10.4).
An endpoint MUST NOT send this frame if it currently requires that
its peer send packets with a zero-length Destination Connection ID.
Changing the length of a connection ID to or from zero-length makes
it difficult to identify when the value of the connection ID changed.
An endpoint that is sending packets with a zero-length Destination
Connection ID MUST treat receipt of a NEW_CONNECTION_ID frame as a
connection error of type PROTOCOL_VIOLATION.
7.14. STOP_SENDING Frame
An endpoint may use a STOP_SENDING frame (type=0x0c) to communicate
that incoming data is being discarded on receipt at application
request. This signals a peer to abruptly terminate transmission on a
stream.
Receipt of a STOP_SENDING frame is only valid for a send stream that
exists and is not in the "Ready" state (see Section 9.2.1).
Receiving a STOP_SENDING frame for a send stream that is "Ready" or
non-existent MUST be treated as a connection error of type
PROTOCOL_VIOLATION. An endpoint that receives a STOP_SENDING frame
for a receive-only stream MUST terminate the connection with error
PROTOCOL_VIOLATION.
The STOP_SENDING frame is as follows:
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The fields in the ACK frame are as follows:
Largest Acknowledged: A variable-length integer representing the
largest packet number the peer is acknowledging; this is usually
the largest packet number that the peer has received prior to
generating the ACK frame. Unlike the packet number in the QUIC
long or short header, the value in an ACK frame is not truncated.
ACK Delay: A variable-length integer including the time in
microseconds that the largest acknowledged packet, as indicated in
the Largest Acknowledged field, was received by this peer to when
this ACK was sent. The value of the ACK Delay field is scaled by
multiplying the encoded value by the 2 to the power of the value
of the "ack_delay_exponent" transport parameter set by the sender
of the ACK frame. The "ack_delay_exponent" defaults to 3, or a
multiplier of 8 (see Section 6.4.1). Scaling in this fashion
allows for a larger range of values with a shorter encoding at the
cost of lower resolution.
ACK Block Count: The number of Additional ACK Block (and Gap) fields
after the First ACK Block.
ACK Blocks: Contains one or more blocks of packet numbers which have
been successfully received, see Section 7.15.1.
7.15.1. ACK Block Section
The ACK Block Section consists of alternating Gap and ACK Block
fields in descending packet number order. A First Ack Block field is
followed by a variable number of alternating Gap and Additional ACK
Blocks. The number of Gap and Additional ACK Block fields is
determined by the ACK Block Count field.
Gap and ACK Block fields use a relative integer encoding for
efficiency. Though each encoded value is positive, the values are
subtracted, so that each ACK Block describes progressively lower-
numbered packets. As long as contiguous ranges of packets are small,
the variable-length integer encoding ensures that each range can be
expressed in a small number of octets.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: ACK Block Section
Each ACK Block acknowledges a contiguous range of packets by
indicating the number of acknowledged packets that precede the
largest packet number in that block. A value of zero indicates that
only the largest packet number is acknowledged. Larger ACK Block
values indicate a larger range, with corresponding lower values for
the smallest packet number in the range. Thus, given a largest
packet number for the ACK, the smallest value is determined by the
formula:
smallest = largest - ack_block
The range of packets that are acknowledged by the ACK block include
the range from the smallest packet number to the largest, inclusive.
The largest value for the First ACK Block is determined by the
Largest Acknowledged field; the largest for Additional ACK Blocks is
determined by cumulatively subtracting the size of all preceding ACK
Blocks and Gaps.
Each Gap indicates a range of packets that are not being
acknowledged. The number of packets in the gap is one higher than
the encoded value of the Gap Field.
The value of the Gap field establishes the largest packet number
value for the ACK block that follows the gap using the following
formula:
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largest = previous_smallest - gap - 2
If the calculated value for largest or smallest packet number for any
ACK Block is negative, an endpoint MUST generate a connection error
of type FRAME_ERROR indicating an error in an ACK frame (that is,
0x10d).
The fields in the ACK Block Section are:
First ACK Block: A variable-length integer indicating the number of
contiguous packets preceding the Largest Acknowledged that are
being acknowledged.
Gap (repeated): A variable-length integer indicating the number of
contiguous unacknowledged packets preceding the packet number one
lower than the smallest in the preceding ACK Block.
ACK Block (repeated): A variable-length integer indicating the
number of contiguous acknowledged packets preceding the largest
packet number, as determined by the preceding Gap.
7.15.2. Sending ACK Frames
Implementations MUST NOT generate packets that only contain ACK
frames in response to packets which only contain ACK frames.
However, they MUST acknowledge packets containing only ACK frames
when sending ACK frames in response to other packets.
Implementations MUST NOT send more than one packet containing only
ACK frames per received packet that contains frames other than ACK
frames. Packets containing non-ACK frames MUST be acknowledged
immediately or when a delayed ack timer expires.
To limit ACK blocks to those that have not yet been received by the
sender, the receiver SHOULD track which ACK frames have been
acknowledged by its peer. Once an ACK frame has been acknowledged,
the packets it acknowledges SHOULD NOT be acknowledged again.
Because ACK frames are not sent in response to ACK-only packets, a
receiver that is only sending ACK frames will only receive
acknowledgements for its packets if the sender includes them in
packets with non-ACK frames. A sender SHOULD bundle ACK frames with
other frames when possible.
To limit receiver state or the size of ACK frames, a receiver MAY
limit the number of ACK blocks it sends. A receiver can do this even
without receiving acknowledgment of its ACK frames, with the
knowledge this could cause the sender to unnecessarily retransmit
some data. Standard QUIC [QUIC-RECOVERY] algorithms declare packets
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lost after sufficiently newer packets are acknowledged. Therefore,
the receiver SHOULD repeatedly acknowledge newly received packets in
preference to packets received in the past.
7.15.3. ACK Frames and Packet Protection
ACK frames that acknowledge protected packets MUST be carried in a
packet that has an equivalent or greater level of packet protection.
Packets that are protected with 1-RTT keys MUST be acknowledged in
packets that are also protected with 1-RTT keys.
A packet that is not protected and claims to acknowledge a packet
number that was sent with packet protection is not valid. An
unprotected packet that carries acknowledgments for protected packets
MUST be discarded in its entirety.
Packets that a client sends with 0-RTT packet protection MUST be
acknowledged by the server in packets protected by 1-RTT keys. This
can mean that the client is unable to use these acknowledgments if
the server cryptographic handshake messages are delayed or lost.
Note that the same limitation applies to other data sent by the
server protected by the 1-RTT keys.
Unprotected packets, such as those that carry the initial
cryptographic handshake messages, MAY be acknowledged in unprotected
packets. Unprotected packets are vulnerable to falsification or
modification. Unprotected packets can be acknowledged along with
protected packets in a protected packet.
An endpoint SHOULD acknowledge packets containing cryptographic
handshake messages in the next unprotected packet that it sends,
unless it is able to acknowledge those packets in later packets
protected by 1-RTT keys. At the completion of the cryptographic
handshake, both peers send unprotected packets containing
cryptographic handshake messages followed by packets protected by
1-RTT keys. An endpoint SHOULD acknowledge the unprotected packets
that complete the cryptographic handshake in a protected packet,
because its peer is guaranteed to have access to 1-RTT packet
protection keys.
For instance, a server acknowledges a TLS ClientHello in the packet
that carries the TLS ServerHello; similarly, a client can acknowledge
a TLS HelloRetryRequest in the packet containing a second TLS
ClientHello. The complete set of server handshake messages (TLS
ServerHello through to Finished) might be acknowledged by a client in
protected packets, because it is certain that the server is able to
decipher the packet.
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Endpoints can use PATH_CHALLENGE frames (type=0x0e) to check
reachability to the peer and for path validation during connection
establishment and connection migration.
PATH_CHALLENGE frames contain an 8-byte payload.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Data (8) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Data: This 8-byte field contains arbitrary data.
A PATH_CHALLENGE frame containing 8 octets that are hard to guess is
sufficient to ensure that it is easier to receive the packet than it
is to guess the value correctly.
The recipient of this frame MUST generate a PATH_RESPONSE frame
(Section 7.17) containing the same Data.
7.17. PATH_RESPONSE Frame
The PATH_RESPONSE frame (type=0x0f) is sent in response to a
PATH_CHALLENGE frame. Its format is identical to the PATH_CHALLENGE
frame (Section 7.16).
If the content of a PATH_RESPONSE frame does not match the content of
a PATH_CHALLENGE frame previously sent by the endpoint, the endpoint
MAY generate a connection error of type UNSOLICITED_PATH_RESPONSE.
7.18. STREAM Frames
STREAM frames implicitly create a stream and carry stream data. The
STREAM frame takes the form 0b00010XXX (or the set of values from
0x10 to 0x17). The value of the three low-order bits of the frame
type determine the fields that are present in the frame.
o The OFF bit (0x04) in the frame type is set to indicate that there
is an Offset field present. When set to 1, the Offset field is
present; when set to 0, the Offset field is absent and the Stream
Data starts at an offset of 0 (that is, the frame contains the
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first octets of the stream, or the end of a stream that includes
no data).
o The LEN bit (0x02) in the frame type is set to indicate that there
is a Length field present. If this bit is set to 0, the Length
field is absent and the Stream Data field extends to the end of
the packet. If this bit is set to 1, the Length field is present.
o The FIN bit (0x01) of the frame type is set only on frames that
contain the final offset of the stream. Setting this bit
indicates that the frame marks the end of the stream.
An endpoint that receives a STREAM frame for a send-only stream MUST
terminate the connection with error PROTOCOL_VIOLATION.
A STREAM frame is shown below.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Offset (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Length (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: STREAM Frame Format
The STREAM frame contains the following fields:
Stream ID: A variable-length integer indicating the stream ID of the
stream (see Section 9.1).
Offset: A variable-length integer specifying the byte offset in the
stream for the data in this STREAM frame. This field is present
when the OFF bit is set to 1. When the Offset field is absent,
the offset is 0.
Length: A variable-length integer specifying the length of the
Stream Data field in this STREAM frame. This field is present
when the LEN bit is set to 1. When the LEN bit is set to 0, the
Stream Data field consumes all the remaining octets in the packet.
Stream Data: The bytes from the designated stream to be delivered.
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When a Stream Data field has a length of 0, the offset in the STREAM
frame is the offset of the next byte that would be sent.
The first byte in the stream has an offset of 0. The largest offset
delivered on a stream - the sum of the re-constructed offset and data
length - MUST be less than 2^62.
Stream multiplexing is achieved by interleaving STREAM frames from
multiple streams into one or more QUIC packets. A single QUIC packet
can include multiple STREAM frames from one or more streams.
Implementation note: One of the benefits of QUIC is avoidance of
head-of-line blocking across multiple streams. When a packet loss
occurs, only streams with data in that packet are blocked waiting for
a retransmission to be received, while other streams can continue
making progress. Note that when data from multiple streams is
bundled into a single QUIC packet, loss of that packet blocks all
those streams from making progress. An implementation is therefore
advised to bundle as few streams as necessary in outgoing packets
without losing transmission efficiency to underfilled packets.
8. Packetization and Reliability
A sender bundles one or more frames in a QUIC packet (see Section 5).
A sender SHOULD minimize per-packet bandwidth and computational costs
by bundling as many frames as possible within a QUIC packet. A
sender MAY wait for a short period of time to bundle multiple frames
before sending a packet that is not maximally packed, to avoid
sending out large numbers of small packets. An implementation may
use knowledge about application sending behavior or heuristics to
determine whether and for how long to wait. This waiting period is
an implementation decision, and an implementation should be careful
to delay conservatively, since any delay is likely to increase
application-visible latency.
8.1. Packet Processing and Acknowledgment
A packet MUST NOT be acknowledged until packet protection has been
successfully removed and all frames contained in the packet have been
processed. Any stream state transitions triggered by the frame MUST
have occurred. For STREAM frames, this means the data has been
enqueued in preparation to be received by the application protocol,
but it does not require that data is delivered and consumed.
Once the packet has been fully processed, a receiver acknowledges
receipt by sending one or more ACK frames containing the packet
number of the received packet. To avoid creating an indefinite
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feedback loop, an endpoint MUST NOT send an ACK frame in response to
a packet containing only ACK or PADDING frames, even if there are
packet gaps which precede the received packet. The endpoint MUST
acknowledge packets containing only ACK or PADDING frames in the next
ACK frame that it sends.
Strategies and implications of the frequency of generating
acknowledgments are discussed in more detail in [QUIC-RECOVERY].
8.2. Retransmission of Information
QUIC packets that are determined to be lost are not retransmitted
whole. The same applies to the frames that are contained within lost
packets. Instead, the information that might be carried in frames is
sent again in new frames as needed.
New frames and packets are used to carry information that is
determined to have been lost. In general, information is sent again
when a packet containing that information is determined to be lost
and sending ceases when a packet containing that information is
acknowledged.
o Application data sent in STREAM frames is retransmitted in new
STREAM frames unless the endpoint has sent a RST_STREAM for that
stream. Once an endpoint sends a RST_STREAM frame, no further
STREAM frames are needed.
o The most recent set of acknowledgments are sent in ACK frames. An
ACK frame SHOULD contain all unacknowledged acknowledgments, as
described in Section 7.15.2.
o Cancellation of stream transmission, as carried in a RST_STREAM
frame, is sent until acknowledged or until all stream data is
acknowledged by the peer (that is, either the "Reset Recvd" or
"Data Recvd" state is reached on the send stream). The content of
a RST_STREAM frame MUST NOT change when it is sent again.
o Similarly, a request to cancel stream transmission, as encoded in
a STOP_SENDING frame, is sent until the receive stream enters
either a "Data Recvd" or "Reset Recvd" state, see Section 9.3.
o Connection close signals, including those that use
CONNECTION_CLOSE and APPLICATION_CLOSE frames, are not sent again
when packet loss is detected, but as described in Section 6.10.
o The current connection maximum data is sent in MAX_DATA frames.
An updated value is sent in a MAX_DATA frame if the packet
containing the most recently sent MAX_DATA frame is declared lost,
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or when the endpoint decides to update the limit. Care is
necessary to avoid sending this frame too often as the limit can
increase frequently and cause an unnecessarily large number of
MAX_DATA frames to be sent.
o The current maximum stream data offset is sent in MAX_STREAM_DATA
frames. Like MAX_DATA, an updated value is sent when the packet
containing the most recent MAX_STREAM_DATA frame for a stream is
lost or when the limit is updated, with care taken to prevent the
frame from being sent too often. An endpoint SHOULD stop sending
MAX_STREAM_DATA frames when the receive stream enters a "Size
Known" state.
o The maximum stream ID for a stream of a given type is sent in
MAX_STREAM_ID frames. Like MAX_DATA, an updated value is sent
when a packet containing the most recent MAX_STREAM_ID for a
stream type frame is declared lost or when the limit is updated,
with care taken to prevent the frame from being sent too often.
o Blocked signals are carried in BLOCKED, STREAM_BLOCKED, and
STREAM_ID_BLOCKED frames. BLOCKED streams have connection scope,
STREAM_BLOCKED frames have stream scope, and STREAM_ID_BLOCKED
frames are scoped to a specific stream type. New frames are sent
if packets containing the most recent frame for a scope is lost,
but only while the endpoint is blocked on the corresponding limit.
These frames always include the limit that is causing blocking at
the time that they are transmitted.
o A liveness or path validation check using PATH_CHALLENGE frames is
sent periodically until a matching PATH_RESPONSE frame is received
or until there is no remaining need for liveness or path
validation checking. PATH_CHALLENGE frames include a different
payload each time they are sent.
o Responses to path validation using PATH_RESPONSE frames are sent
just once. A new PATH_CHALLENGE frame will be sent if another
PATH_RESPONSE frame is needed.
o New connection IDs are sent in NEW_CONNECTION_ID frames and
retransmitted if the packet containing them is lost.
o PADDING frames contain no information, so lost PADDING frames do
not require repair.
Upon detecting losses, a sender MUST take appropriate congestion
control action. The details of loss detection and congestion control
are described in [QUIC-RECOVERY].
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The QUIC packet size includes the QUIC header and integrity check,
but not the UDP or IP header.
Clients MUST pad any Initial packet it sends to have a QUIC packet
size of at least 1200 octets. Sending an Initial packet of this size
ensures that the network path supports a reasonably sized packet, and
helps reduce the amplitude of amplification attacks caused by server
responses toward an unverified client address.
An Initial packet MAY exceed 1200 octets if the client knows that the
Path Maximum Transmission Unit (PMTU) supports the size that it
chooses.
A server MAY send a CONNECTION_CLOSE frame with error code
PROTOCOL_VIOLATION in response to an Initial packet smaller than 1200
octets. It MUST NOT send any other frame type in response, or
otherwise behave as if any part of the offending packet was processed
as valid.
8.4. Path Maximum Transmission Unit
The Path Maximum Transmission Unit (PMTU) is the maximum size of the
entire IP header, UDP header, and UDP payload. The UDP payload
includes the QUIC packet header, protected payload, and any
authentication fields.
All QUIC packets SHOULD be sized to fit within the estimated PMTU to
avoid IP fragmentation or packet drops. To optimize bandwidth
efficiency, endpoints SHOULD use Packetization Layer PMTU Discovery
([PLPMTUD]). Endpoints MAY use PMTU Discovery ([PMTUDv4], [PMTUDv6])
for detecting the PMTU, setting the PMTU appropriately, and storing
the result of previous PMTU determinations.
In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP
packets larger than 1280 octets. Assuming the minimum IP header
size, this results in a QUIC packet size of 1232 octets for IPv6 and
1252 octets for IPv4. Some QUIC implementations MAY wish to be more
conservative in computing allowed QUIC packet size given unknown
tunneling overheads or IP header options.
QUIC endpoints that implement any kind of PMTU discovery SHOULD
maintain an estimate for each combination of local and remote IP
addresses. Each pairing of local and remote addresses could have a
different maximum MTU in the path.
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QUIC depends on the network path supporting a MTU of at least 1280
octets. This is the IPv6 minimum MTU and therefore also supported by
most modern IPv4 networks. An endpoint MUST NOT reduce its MTU below
this number, even if it receives signals that indicate a smaller
limit might exist.
If a QUIC endpoint determines that the PMTU between any pair of local
and remote IP addresses has fallen below 1280 octets, it MUST
immediately cease sending QUIC packets on the affected path. This
could result in termination of the connection if an alternative path
cannot be found.
8.4.1. Special Considerations for PMTU Discovery
Traditional ICMP-based path MTU discovery in IPv4 [PMTUDv4] is
potentially vulnerable to off-path attacks that successfully guess
the IP/port 4-tuple and reduce the MTU to a bandwidth-inefficient
value. TCP connections mitigate this risk by using the (at minimum)
8 bytes of transport header echoed in the ICMP message to validate
the TCP sequence number as valid for the current connection.
However, as QUIC operates over UDP, in IPv4 the echoed information
could consist only of the IP and UDP headers, which usually has
insufficient entropy to mitigate off-path attacks.
As a result, endpoints that implement PMTUD in IPv4 SHOULD take steps
to mitigate this risk. For instance, an application could:
o Set the IPv4 Don't Fragment (DF) bit on a small proportion of
packets, so that most invalid ICMP messages arrive when there are
no DF packets outstanding, and can therefore be identified as
spurious.
o Store additional information from the IP or UDP headers from DF
packets (for example, the IP ID or UDP checksum) to further
authenticate incoming Datagram Too Big messages.
o Any reduction in PMTU due to a report contained in an ICMP packet
is provisional until QUIC's loss detection algorithm determines
that the packet is actually lost.
8.4.2. Special Considerations for Packetization Layer PMTU Discovery
The PADDING frame provides a useful option for PMTU probe packets.
PADDING frames generate acknowledgements, but they need not be
delivered reliably. As a result, the loss of PADDING frames in probe
packets does not require delay-inducing retransmission. However,
PADDING frames do consume congestion window, which may delay the
transmission of subsequent application data.
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When implementing the algorithm in Section 7.2 of [PLPMTUD], the
initial value of search_low SHOULD be consistent with the IPv6
minimum packet size. Paths that do not support this size cannot
deliver Initial packets, and therefore are not QUIC-compliant.
Section 7.3 of [PLPMTUD] discusses tradeoffs between small and large
increases in the size of probe packets. As QUIC probe packets need
not contain application data, aggressive increases in probe size
carry fewer consequences.
9. Streams: QUIC's Data Structuring Abstraction
Streams in QUIC provide a lightweight, ordered byte-stream
abstraction.
There are two basic types of stream in QUIC. Unidirectional streams
carry data in one direction only; bidirectional streams allow for
data to be sent in both directions. Different stream identifiers are
used to distinguish between unidirectional and bidirectional streams,
as well as to create a separation between streams that are initiated
by the client and server (see Section 9.1).
Either type of stream can be created by either endpoint, can
concurrently send data interleaved with other streams, and can be
cancelled.
Stream offsets allow for the octets on a stream to be placed in
order. An endpoint MUST be capable of delivering data received on a
stream in order. Implementations MAY choose to offer the ability to
deliver data out of order. There is no means of ensuring ordering
between octets on different streams.
The creation and destruction of streams are expected to have minimal
bandwidth and computational cost. A single STREAM frame may create,
carry data for, and terminate a stream, or a stream may last the
entire duration of a connection.
Streams are individually flow controlled, allowing an endpoint to
limit memory commitment and to apply back pressure. The creation of
streams is also flow controlled, with each peer declaring the maximum
stream ID it is willing to accept at a given time.
An alternative view of QUIC streams is as an elastic "message"
abstraction, similar to the way ephemeral streams are used in SST
[SST], which may be a more appealing description for some
applications.
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Streams are identified by an unsigned 62-bit integer, referred to as
the Stream ID. The least significant two bits of the Stream ID are
used to identify the type of stream (unidirectional or bidirectional)
and the initiator of the stream.
The least significant bit (0x1) of the Stream ID identifies the
initiator of the stream. Clients initiate even-numbered streams
(those with the least significant bit set to 0); servers initiate
odd-numbered streams (with the bit set to 1). Separation of the
stream identifiers ensures that client and server are able to open
streams without the latency imposed by negotiating for an identifier.
If an endpoint receives a frame for a stream that it expects to
initiate (i.e., odd-numbered for the client or even-numbered for the
server), but which it has not yet opened, it MUST close the
connection with error code STREAM_STATE_ERROR.
The second least significant bit (0x2) of the Stream ID
differentiates between unidirectional streams and bidirectional
streams. Unidirectional streams always have this bit set to 1 and
bidirectional streams have this bit set to 0.
The two type bits from a Stream ID therefore identify streams as
summarized in Table 5.
+----------+----------------------------------+
| Low Bits | Stream Type |
+----------+----------------------------------+
| 0x0 | Client-Initiated, Bidirectional |
| | |
| 0x1 | Server-Initiated, Bidirectional |
| | |
| 0x2 | Client-Initiated, Unidirectional |
| | |
| 0x3 | Server-Initiated, Unidirectional |
+----------+----------------------------------+
Table 5: Stream ID Types
Stream ID 0 (0x0) is a client-initiated, bidirectional stream that is
used for the cryptographic handshake. Stream 0 MUST NOT be used for
application data.
A QUIC endpoint MUST NOT reuse a Stream ID. Streams can be used in
any order. Streams that are used out of order result in opening all
lower-numbered streams of the same type in the same direction.
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Stream IDs are encoded as a variable-length integer (see
Section 7.1).
9.2. Stream States
This section describes the two types of QUIC stream in terms of the
states of their send or receive components. Two state machines are
described: one for streams on which an endpoint transmits data
(Section 9.2.1); another for streams from which an endpoint receives
data (Section 9.2.2).
Unidirectional streams use the applicable state machine directly.
Bidirectional streams use both state machines. For the most part,
the use of these state machines is the same whether the stream is
unidirectional or bidirectional. The conditions for opening a stream
are slightly more complex for a bidirectional stream because the
opening of either send or receive sides causes the stream to open in
both directions.
An endpoint can open streams up to its maximum stream limit in any
order, however endpoints SHOULD open the send side of streams for
each type in order.
Note: These states are largely informative. This document uses
stream states to describe rules for when and how different types
of frames can be sent and the reactions that are expected when
different types of frames are received. Though these state
machines are intended to be useful in implementing QUIC, these
states aren't intended to constrain implementations. An
implementation can define a different state machine as long as its
behavior is consistent with an implementation that implements
these states.
9.2.1. Send Stream States
Figure 10 shows the states for the part of a stream that sends data
to a peer.
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In the "Send" state, an endpoint transmits - and retransmits as
necessary - data in STREAM frames. The endpoint respects the flow
control limits of its peer, accepting MAX_STREAM_DATA frames. An
endpoint in the "Send" state generates STREAM_BLOCKED frames if it
encounters flow control limits.
After the application indicates that stream data is complete and a
STREAM frame containing the FIN bit is sent, the send stream enters
the "Data Sent" state. From this state, the endpoint only
retransmits stream data as necessary. The endpoint no longer needs
to track flow control limits or send STREAM_BLOCKED frames for a send
stream in this state. The endpoint can ignore any MAX_STREAM_DATA
frames it receives from its peer in this state; MAX_STREAM_DATA
frames might be received until the peer receives the final stream
offset.
Once all stream data has been successfully acknowledged, the send
stream enters the "Data Recvd" state, which is a terminal state.
From any of the "Ready", "Send", or "Data Sent" states, an
application can signal that it wishes to abandon transmission of
stream data. Similarly, the endpoint might receive a STOP_SENDING
frame from its peer. In either case, the endpoint sends a RST_STREAM
frame, which causes the stream to enter the "Reset Sent" state.
An endpoint MAY send a RST_STREAM as the first frame on a send
stream; this causes the send stream to open and then immediately
transition to the "Reset Sent" state.
Once a packet containing a RST_STREAM has been acknowledged, the send
stream enters the "Reset Recvd" state, which is a terminal state.
9.2.2. Receive Stream States
Figure 11 shows the states for the part of a stream that receives
data from a peer. The states for a receive stream mirror only some
of the states of the send stream at the peer. A receive stream
doesn't track states on the send stream that cannot be observed, such
as the "Ready" state; instead, receive streams track the delivery of
data to the application or application protocol some of which cannot
be observed by the sender.
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MAX_STREAM_DATA frame might arrive before a STREAM or STREAM_BLOCKED
frame if packets are lost or reordered.
In the "Recv" state, the endpoint receives STREAM and STREAM_BLOCKED
frames. Incoming data is buffered and can be reassembled into the
correct order for delivery to the application. As data is consumed
by the application and buffer space becomes available, the endpoint
sends MAX_STREAM_DATA frames to allow the peer to send more data.
When a STREAM frame with a FIN bit is received, the final offset (see
Section 10.3) is known. The receive stream enters the "Size Known"
state. In this state, the endpoint no longer needs to send
MAX_STREAM_DATA frames, it only receives any retransmissions of
stream data.
Once all data for the stream has been received, the receive stream
enters the "Data Recvd" state. This might happen as a result of
receiving the same STREAM frame that causes the transition to "Size
Known". In this state, the endpoint has all stream data. Any STREAM
or STREAM_BLOCKED frames it receives for the stream can be discarded.
The "Data Recvd" state persists until stream data has been delivered
to the application or application protocol. Once stream data has
been delivered, the stream enters the "Data Read" state, which is a
terminal state.
Receiving a RST_STREAM frame in the "Recv" or "Size Known" states
causes the stream to enter the "Reset Recvd" state. This might cause
the delivery of stream data to the application to be interrupted.
It is possible that all stream data is received when a RST_STREAM is
received (that is, from the "Data Recvd" state). Similarly, it is
possible for remaining stream data to arrive after receiving a
RST_STREAM frame (the "Reset Recvd" state). An implementation is
able to manage this situation as they choose. Sending RST_STREAM
means that an endpoint cannot guarantee delivery of stream data;
however there is no requirement that stream data not be delivered if
a RST_STREAM is received. An implementation MAY interrupt delivery
of stream data, discard any data that was not consumed, and signal
the existence of the RST_STREAM immediately. Alternatively, the
RST_STREAM signal might be suppressed or withheld if stream data is
completely received. In the latter case, the receive stream
effectively transitions to "Data Recvd" from "Reset Recvd".
Once the application has been delivered the signal indicating that
the receive stream was reset, the receive stream transitions to the
"Reset Read" state, which is a terminal state.
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The sender of a stream sends just three frame types that affect the
state of a stream at either sender or receiver: STREAM
(Section 7.18), STREAM_BLOCKED (Section 7.11), and RST_STREAM
(Section 7.3).
A sender MUST NOT send any of these frames from a terminal state
("Data Recvd" or "Reset Recvd"). A sender MUST NOT send STREAM or
STREAM_BLOCKED after sending a RST_STREAM; that is, in the "Reset
Sent" state in addition to the terminal states. A receiver could
receive any of these frames in any state, but only due to the
possibility of delayed delivery of packets carrying them.
The receiver of a stream sends MAX_STREAM_DATA (Section 7.7) and
STOP_SENDING frames (Section 7.14).
The receiver only sends MAX_STREAM_DATA in the "Recv" state. A
receiver can send STOP_SENDING in any state where it has not received
a RST_STREAM frame; that is states other than "Reset Recvd" or "Reset
Read". However there is little value in sending a STOP_SENDING frame
after all stream data has been received in the "Data Recvd" state. A
sender could receive these frames in any state as a result of delayed
delivery of packets.
9.2.4. Bidirectional Stream States
A bidirectional stream is composed of a send stream and a receive
stream. Implementations may represent states of the bidirectional
stream as composites of send and receive stream states. The simplest
model presents the stream as "open" when either send or receive
stream is in a non-terminal state and "closed" when both send and
receive streams are in a terminal state.
Table 6 shows a more complex mapping of bidirectional stream states
that loosely correspond to the stream states in HTTP/2 [HTTP2]. This
shows that multiple states on send or receive streams are mapped to
the same composite state. Note that this is just one possibility for
such a mapping; this mapping requires that data is acknowledged
before the transition to a "closed" or "half-closed" state.
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STREAM frames received after sending STOP_SENDING are still counted
toward the connection and stream flow-control windows, even though
these frames will be discarded upon receipt. This avoids potential
ambiguity about which STREAM frames count toward flow control.
A STOP_SENDING frame requests that the receiving endpoint send a
RST_STREAM frame. An endpoint that receives a STOP_SENDING frame
MUST send a RST_STREAM frame for that stream, and can use an error
code of STOPPING. If the STOP_SENDING frame is received on a send
stream that is already in the "Data Sent" state, a RST_STREAM frame
MAY still be sent in order to cancel retransmission of previously-
sent STREAM frames.
STOP_SENDING SHOULD only be sent for a receive stream that has not
been reset. STOP_SENDING is most useful for streams in the "Recv" or
"Size Known" states.
An endpoint is expected to send another STOP_SENDING frame if a
packet containing a previous STOP_SENDING is lost. However, once
either all stream data or a RST_STREAM frame has been received for
the stream - that is, the stream is in any state other than "Recv" or
"Size Known" - sending a STOP_SENDING frame is unnecessary.
9.4. Stream Concurrency
An endpoint limits the number of concurrently active incoming streams
by adjusting the maximum stream ID. An initial value is set in the
transport parameters (see Section 6.4.1) and is subsequently
increased by MAX_STREAM_ID frames (see Section 7.8).
The maximum stream ID is specific to each endpoint and applies only
to the peer that receives the setting. That is, clients specify the
maximum stream ID the server can initiate, and servers specify the
maximum stream ID the client can initiate. Each endpoint may respond
on streams initiated by the other peer, regardless of whether it is
permitted to initiated new streams.
Endpoints MUST NOT exceed the limit set by their peer. An endpoint
that receives a STREAM frame with an ID greater than the limit it has
sent MUST treat this as a stream error of type STREAM_ID_ERROR
(Section 11), unless this is a result of a change in the initial
offsets (see Section 6.4.2).
A receiver MUST NOT renege on an advertisement; that is, once a
receiver advertises a stream ID via a MAX_STREAM_ID frame, it MUST
NOT subsequently advertise a smaller maximum ID. A sender may
receive MAX_STREAM_ID frames out of order; a sender MUST therefore
ignore any MAX_STREAM_ID that does not increase the maximum.
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Once a stream is created, endpoints may use the stream to send and
receive data. Each endpoint may send a series of STREAM frames
encapsulating data on a stream until the stream is terminated in that
direction. Streams are an ordered byte-stream abstraction, and they
have no other structure within them. STREAM frame boundaries are not
expected to be preserved in retransmissions from the sender or during
delivery to the application at the receiver.
When new data is to be sent on a stream, a sender MUST set the
encapsulating STREAM frame's offset field to the stream offset of the
first byte of this new data. The first octet of data on a stream has
an offset of 0. An endpoint is expected to send every stream octet.
The largest offset delivered on a stream MUST be less than 2^62.
QUIC makes no specific allowances for partial reliability or delivery
of stream data out of order. Endpoints MUST be able to deliver
stream data to an application as an ordered byte-stream. Delivering
an ordered byte-stream requires that an endpoint buffer any data that
is received out of order, up to the advertised flow control limit.
An endpoint could receive the same octets multiple times; octets that
have already been received can be discarded. The value for a given
octet MUST NOT change if it is sent multiple times; an endpoint MAY
treat receipt of a changed octet as a connection error of type
PROTOCOL_VIOLATION.
An endpoint MUST NOT send data on any stream without ensuring that it
is within the data limits set by its peer. The cryptographic
handshake stream, Stream 0, is exempt from the connection-level data
limits established by MAX_DATA. Data on stream 0 other than the
initial cryptographic handshake message is still subject to stream-
level data limits and MAX_STREAM_DATA. This message is exempt from
flow control because it needs to be sent in a single packet
regardless of the server's flow control state. This rule applies
even for 0-RTT handshakes where the remembered value of
MAX_STREAM_DATA would not permit sending a full initial cryptographic
handshake message.
Flow control is described in detail in Section 10, and congestion
control is described in the companion document [QUIC-RECOVERY].
9.6. Stream Prioritization
Stream multiplexing has a significant effect on application
performance if resources allocated to streams are correctly
prioritized. Experience with other multiplexed protocols, such as
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HTTP/2 [HTTP2], shows that effective prioritization strategies have a
significant positive impact on performance.
QUIC does not provide frames for exchanging prioritization
information. Instead it relies on receiving priority information
from the application that uses QUIC. Protocols that use QUIC are
able to define any prioritization scheme that suits their application
semantics. A protocol might define explicit messages for signaling
priority, such as those defined in HTTP/2; it could define rules that
allow an endpoint to determine priority based on context; or it could
leave the determination to the application.
A QUIC implementation SHOULD provide ways in which an application can
indicate the relative priority of streams. When deciding which
streams to dedicate resources to, QUIC SHOULD use the information
provided by the application. Failure to account for priority of
streams can result in suboptimal performance.
Stream priority is most relevant when deciding which stream data will
be transmitted. Often, there will be limits on what can be
transmitted as a result of connection flow control or the current
congestion controller state.
Giving preference to the transmission of its own management frames
ensures that the protocol functions efficiently. That is,
prioritizing frames other than STREAM frames ensures that loss
recovery, congestion control, and flow control operate effectively.
Stream 0 MUST be prioritized over other streams prior to the
completion of the cryptographic handshake. This includes the
retransmission of the second flight of client handshake messages,
that is, the TLS Finished and any client authentication messages.
STREAM data in frames determined to be lost SHOULD be retransmitted
before sending new data, unless application priorities indicate
otherwise. Retransmitting lost stream data can fill in gaps, which
allows the peer to consume already received data and free up flow
control window.
10. Flow Control
It is necessary to limit the amount of data that a sender may have
outstanding at any time, so as to prevent a fast sender from
overwhelming a slow receiver, or to prevent a malicious sender from
consuming significant resources at a receiver. This section
describes QUIC's flow-control mechanisms.
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QUIC employs a credit-based flow-control scheme similar to HTTP/2's
flow control [HTTP2]. A receiver advertises the number of octets it
is prepared to receive on a given stream and for the entire
connection. This leads to two levels of flow control in QUIC: (i)
Connection flow control, which prevents senders from exceeding a
receiver's buffer capacity for the connection, and (ii) Stream flow
control, which prevents a single stream from consuming the entire
receive buffer for a connection.
A data receiver sends MAX_STREAM_DATA or MAX_DATA frames to the
sender to advertise additional credit. MAX_STREAM_DATA frames send
the the maximum absolute byte offset of a stream, while MAX_DATA
sends the maximum sum of the absolute byte offsets of all streams
other than stream 0.
A receiver MAY advertise a larger offset at any point by sending
MAX_DATA or MAX_STREAM_DATA frames. A receiver MUST NOT renege on an
advertisement; that is, once a receiver advertises an offset, it MUST
NOT subsequently advertise a smaller offset. A sender could receive
MAX_DATA or MAX_STREAM_DATA frames out of order; a sender MUST
therefore ignore any flow control offset that does not move the
window forward.
A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
(Section 11) if the peer violates the advertised connection or stream
data limits.
A sender SHOULD send BLOCKED or STREAM_BLOCKED frames to indicate it
has data to write but is blocked by flow control limits. These
frames are expected to be sent infrequently in common cases, but they
are considered useful for debugging and monitoring purposes.
A receiver advertises credit for a stream by sending a
MAX_STREAM_DATA frame with the Stream ID set appropriately. A
receiver could use the current offset of data consumed to determine
the flow control offset to be advertised. A receiver MAY send
MAX_STREAM_DATA frames in multiple packets in order to make sure that
the sender receives an update before running out of flow control
credit, even if one of the packets is lost.
Connection flow control is a limit to the total bytes of stream data
sent in STREAM frames on all streams except stream 0. A receiver
advertises credit for a connection by sending a MAX_DATA frame. A
receiver maintains a cumulative sum of bytes received on all
contributing streams, which are used to check for flow control
violations. A receiver might use a sum of bytes consumed on all
contributing streams to determine the maximum data limit to be
advertised.
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Internet-Draft QUIC Transport Protocol May 201810.1. Edge Cases and Other Considerations
There are some edge cases which must be considered when dealing with
stream and connection level flow control. Given enough time, both
endpoints must agree on flow control state. If one end believes it
can send more than the other end is willing to receive, the
connection will be torn down when too much data arrives.
Conversely if a sender believes it is blocked, while endpoint B
expects more data can be received, then the connection can be in a
deadlock, with the sender waiting for a MAX_DATA or MAX_STREAM_DATA
frame which will never come.
On receipt of a RST_STREAM frame, an endpoint will tear down state
for the matching stream and ignore further data arriving on that
stream. This could result in the endpoints getting out of sync,
since the RST_STREAM frame may have arrived out of order and there
may be further bytes in flight. The data sender would have counted
the data against its connection level flow control budget, but a
receiver that has not received these bytes would not know to include
them as well. The receiver must learn the number of bytes that were
sent on the stream to make the same adjustment in its connection flow
controller.
To avoid this de-synchronization, a RST_STREAM sender MUST include
the final byte offset sent on the stream in the RST_STREAM frame. On
receiving a RST_STREAM frame, a receiver definitively knows how many
bytes were sent on that stream before the RST_STREAM frame, and the
receiver MUST use the final offset to account for all bytes sent on
the stream in its connection level flow controller.
10.1.1. Response to a RST_STREAM
RST_STREAM terminates one direction of a stream abruptly. Whether
any action or response can or should be taken on the data already
received is an application-specific issue, but it will often be the
case that upon receipt of a RST_STREAM an endpoint will choose to
stop sending data in its own direction. If the sender of a
RST_STREAM wishes to explicitly state that no future data will be
processed, that endpoint MAY send a STOP_SENDING frame at the same
time.
10.1.2. Data Limit Increments
This document leaves when and how many bytes to advertise in a
MAX_DATA or MAX_STREAM_DATA to implementations, but offers a few
considerations. These frames contribute to connection overhead.
Therefore frequently sending frames with small changes is
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undesirable. At the same time, infrequent updates require larger
increments to limits if blocking is to be avoided. Thus, larger
updates require a receiver to commit to larger resource commitments.
Thus there is a tradeoff between resource commitment and overhead
when determining how large a limit is advertised.
A receiver MAY use an autotuning mechanism to tune the frequency and
amount that it increases data limits based on a round-trip time
estimate and the rate at which the receiving application consumes
data, similar to common TCP implementations.
10.1.3. Handshake Exemption
During the initial handshake, an endpoint could need to send a larger
message on stream 0 than would ordinarily be permitted by the peer's
initial stream flow control window. Since MAX_STREAM_DATA frames are
not permitted in these early packets, the peer cannot provide
additional flow control window in order to complete the handshake.
Endpoints MAY exceed the flow control limits on stream 0 prior to the
completion of the cryptographic handshake. (That is, in Initial,
Retry, and Handshake packets.) However, once the handshake is
complete, endpoints MUST NOT send additional data beyond the peer's
permitted offset. If the amount of data sent during the handshake
exceeds the peer's maximum offset, the endpoint cannot send
additional data on stream 0 until the peer has sent a MAX_STREAM_DATA
frame indicating a larger maximum offset.
10.2. Stream Limit Increment
As with flow control, this document leaves when and how many streams
to make available to a peer via MAX_STREAM_ID to implementations, but
offers a few considerations. MAX_STREAM_ID frames constitute minimal
overhead, while withholding MAX_STREAM_ID frames can prevent the peer
from using the available parallelism.
Implementations will likely want to increase the maximum stream ID as
peer-initiated streams close. A receiver MAY also advance the
maximum stream ID based on current activity, system conditions, and
other environmental factors.
10.2.1. Blocking on Flow Control
If a sender does not receive a MAX_DATA or MAX_STREAM_DATA frame when
it has run out of flow control credit, the sender will be blocked and
SHOULD send a BLOCKED or STREAM_BLOCKED frame. These frames are
expected to be useful for debugging at the receiver; they do not
require any other action. A receiver SHOULD NOT wait for a BLOCKED
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or STREAM_BLOCKED frame before sending MAX_DATA or MAX_STREAM_DATA,
since doing so will mean that a sender is unable to send for an
entire round trip.
For smooth operation of the congestion controller, it is generally
considered best to not let the sender go into quiescence if
avoidable. To avoid blocking a sender, and to reasonably account for
the possibiity of loss, a receiver should send a MAX_DATA or
MAX_STREAM_DATA frame at least two round trips before it expects the
sender to get blocked.
A sender sends a single BLOCKED or STREAM_BLOCKED frame only once
when it reaches a data limit. A sender SHOULD NOT send multiple
BLOCKED or STREAM_BLOCKED frames for the same data limit, unless the
original frame is determined to be lost. Another BLOCKED or
STREAM_BLOCKED frame can be sent after the data limit is increased.
10.3. Stream Final Offset
The final offset is the count of the number of octets that are
transmitted on a stream. For a stream that is reset, the final
offset is carried explicitly in a RST_STREAM frame. Otherwise, the
final offset is the offset of the end of the data carried in a STREAM
frame marked with a FIN flag, or 0 in the case of incoming
unidirectional streams.
An endpoint will know the final offset for a stream when the receive
stream enters the "Size Known" or "Reset Recvd" state.
An endpoint MUST NOT send data on a stream at or beyond the final
offset.
Once a final offset for a stream is known, it cannot change. If a
RST_STREAM or STREAM frame causes the final offset to change for a
stream, an endpoint SHOULD respond with a FINAL_OFFSET_ERROR error
(see Section 11). A receiver SHOULD treat receipt of data at or
beyond the final offset as a FINAL_OFFSET_ERROR error, even after a
stream is closed. Generating these errors is not mandatory, but only
because requiring that an endpoint generate these errors also means
that the endpoint needs to maintain the final offset state for closed
streams, which could mean a significant state commitment.
11. Error Handling
An endpoint that detects an error SHOULD signal the existence of that
error to its peer. Both transport-level and application-level errors
can affect an entire connection (see Section 11.1), while only
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application-level errors can be isolated to a single stream (see
Section 11.2).
The most appropriate error code (Section 11.3) SHOULD be included in
the frame that signals the error. Where this specification
identifies error conditions, it also identifies the error code that
is used.
A stateless reset (Section 6.10.4) is not suitable for any error that
can be signaled with a CONNECTION_CLOSE, APPLICATION_CLOSE, or
RST_STREAM frame. A stateless reset MUST NOT be used by an endpoint
that has the state necessary to send a frame on the connection.
11.1. Connection Errors
Errors that result in the connection being unusable, such as an
obvious violation of protocol semantics or corruption of state that
affects an entire connection, MUST be signaled using a
CONNECTION_CLOSE or APPLICATION_CLOSE frame (Section 7.4,
Section 7.5). An endpoint MAY close the connection in this manner
even if the error only affects a single stream.
Application protocols can signal application-specific protocol errors
using the APPLICATION_CLOSE frame. Errors that are specific to the
transport, including all those described in this document, are
carried in a CONNECTION_CLOSE frame. Other than the type of error
code they carry, these frames are identical in format and semantics.
A CONNECTION_CLOSE or APPLICATION_CLOSE frame could be sent in a
packet that is lost. An endpoint SHOULD be prepared to retransmit a
packet containing either frame type if it receives more packets on a
terminated connection. Limiting the number of retransmissions and
the time over which this final packet is sent limits the effort
expended on terminated connections.
An endpoint that chooses not to retransmit packets containing
CONNECTION_CLOSE or APPLICATION_CLOSE risks a peer missing the first
such packet. The only mechanism available to an endpoint that
continues to receive data for a terminated connection is to use the
stateless reset process (Section 6.10.4).
An endpoint that receives an invalid CONNECTION_CLOSE or
APPLICATION_CLOSE frame MUST NOT signal the existence of the error to
its peer.
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Internet-Draft QUIC Transport Protocol May 201811.2. Stream Errors
If an application-level error affects a single stream, but otherwise
leaves the connection in a recoverable state, the endpoint can send a
RST_STREAM frame (Section 7.3) with an appropriate error code to
terminate just the affected stream.
Stream 0 is critical to the functioning of the entire connection. If
stream 0 is closed with either a RST_STREAM or STREAM frame bearing
the FIN flag, an endpoint MUST generate a connection error of type
PROTOCOL_VIOLATION.
Other than STOPPING (Section 9.3), RST_STREAM MUST be instigated by
the application and MUST carry an application error code. Resetting
a stream without knowledge of the application protocol could cause
the protocol to enter an unrecoverable state. Application protocols
might require certain streams to be reliably delivered in order to
guarantee consistent state between endpoints.
11.3. Transport Error Codes
QUIC error codes are 16-bit unsigned integers.
This section lists the defined QUIC transport error codes that may be
used in a CONNECTION_CLOSE frame. These errors apply to the entire
connection.
NO_ERROR (0x0): An endpoint uses this with CONNECTION_CLOSE to
signal that the connection is being closed abruptly in the absence
of any error.
INTERNAL_ERROR (0x1): The endpoint encountered an internal error and
cannot continue with the connection.
SERVER_BUSY (0x2): The server is currently busy and does not accept
any new connections.
FLOW_CONTROL_ERROR (0x3): An endpoint received more data than it
permitted in its advertised data limits (see Section 10).
STREAM_ID_ERROR (0x4): An endpoint received a frame for a stream
identifier that exceeded its advertised maximum stream ID.
STREAM_STATE_ERROR (0x5): An endpoint received a frame for a stream
that was not in a state that permitted that frame (see
Section 9.2).
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FINAL_OFFSET_ERROR (0x6): An endpoint received a STREAM frame
containing data that exceeded the previously established final
offset. Or an endpoint received a RST_STREAM frame containing a
final offset that was lower than the maximum offset of data that
was already received. Or an endpoint received a RST_STREAM frame
containing a different final offset to the one already
established.
FRAME_FORMAT_ERROR (0x7): An endpoint received a frame that was
badly formatted. For instance, an empty STREAM frame that omitted
the FIN flag, or an ACK frame that has more acknowledgment ranges
than the remainder of the packet could carry. This is a generic
error code; an endpoint SHOULD use the more specific frame format
error codes (0x1XX) if possible.
TRANSPORT_PARAMETER_ERROR (0x8): An endpoint received transport
parameters that were badly formatted, included an invalid value,
was absent even though it is mandatory, was present though it is
forbidden, or is otherwise in error.
VERSION_NEGOTIATION_ERROR (0x9): An endpoint received transport
parameters that contained version negotiation parameters that
disagreed with the version negotiation that it performed. This
error code indicates a potential version downgrade attack.
PROTOCOL_VIOLATION (0xA): An endpoint detected an error with
protocol compliance that was not covered by more specific error
codes.
UNSOLICITED_PATH_RESPONSE (0xB): An endpoint received a
PATH_RESPONSE frame that did not correspond to any PATH_CHALLENGE
frame that it previously sent.
FRAME_ERROR (0x1XX): An endpoint detected an error in a specific
frame type. The frame type is included as the last octet of the
error code. For example, an error in a MAX_STREAM_ID frame would
be indicated with the code (0x106).
Codes for errors occuring when TLS is used for the crypto handshake
are defined in Section 11 of [QUIC-TLS]. See Section 13.2 for
details of registering new error codes.
11.4. Application Protocol Error Codes
Application protocol error codes are 16-bit unsigned integers, but
the management of application error codes are left to application
protocols. Application protocol error codes are used for the
RST_STREAM (Section 7.3) and APPLICATION_CLOSE (Section 7.5) frames.
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There is no restriction on the use of the 16-bit error code space for
application protocols. However, QUIC reserves the error code with a
value of 0 to mean STOPPING. The application error code of STOPPING
(0) is used by the transport to cancel a stream in response to
receipt of a STOP_SENDING frame.
12. Security Considerations12.1. Spoofed ACK Attack
An attacker might be able to receive an address validation token
(Section 6.6) from the server and then release the IP address it used
to acquire that token. The attacker may, in the future, spoof this
same address (which now presumably addresses a different endpoint),
and initiate a 0-RTT connection with a server on the victim's behalf.
The attacker can then spoof ACK frames to the server which cause the
server to send excessive amounts of data toward the new owner of the
IP address.
There are two possible mitigations to this attack. The simplest one
is that a server can unilaterally create a gap in packet-number
space. In the non-attack scenario, the client will send an ACK frame
with the larger value for largest acknowledged. In the attack
scenario, the attacker could acknowledge a packet in the gap. If the
server sees an acknowledgment for a packet that was never sent, the
connection can be aborted.
The second mitigation is that the server can require that
acknowledgments for sent packets match the encryption level of the
sent packet. This mitigation is useful if the connection has an
ephemeral forward-secure key that is generated and used for every new
connection. If a packet sent is protected with a forward-secure key,
then any acknowledgments that are received for them MUST also be
forward-secure protected. Since the attacker will not have the
forward secure key, the attacker will not be able to generate
forward-secure protected packets with ACK frames.
12.2. Optimistic ACK Attack
An endpoint that acknowledges packets it has not received might cause
a congestion controller to permit sending at rates beyond what the
network supports. An endpoint MAY skip packet numbers when sending
packets to detect this behavior. An endpoint can then immediately
close the connection with a connection error of type
PROTOCOL_VIOLATION (see Section 6.10.3).
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Internet-Draft QUIC Transport Protocol May 201812.3. Slowloris Attacks
The attacks commonly known as Slowloris [SLOWLORIS] try to keep many
connections to the target endpoint open and hold them open as long as
possible. These attacks can be executed against a QUIC endpoint by
generating the minimum amount of activity necessary to avoid being
closed for inactivity. This might involve sending small amounts of
data, gradually opening flow control windows in order to control the
sender rate, or manufacturing ACK frames that simulate a high loss
rate.
QUIC deployments SHOULD provide mitigations for the Slowloris
attacks, such as increasing the maximum number of clients the server
will allow, limiting the number of connections a single IP address is
allowed to make, imposing restrictions on the minimum transfer speed
a connection is allowed to have, and restricting the length of time
an endpoint is allowed to stay connected.
12.4. Stream Fragmentation and Reassembly Attacks
An adversarial endpoint might intentionally fragment the data on
stream buffers in order to cause disproportionate memory commitment.
An adversarial endpoint could open a stream and send some STREAM
frames containing arbitrary fragments of the stream content.
The attack is mitigated if flow control windows correspond to
available memory. However, some receivers will over-commit memory
and advertise flow control offsets in the aggregate that exceed
actual available memory. The over-commitment strategy can lead to
better performance when endpoints are well behaved, but renders
endpoints vulnerable to the stream fragmentation attack.
QUIC deployments SHOULD provide mitigations against the stream
fragmentation attack. Mitigations could consist of avoiding over-
committing memory, delaying reassembly of STREAM frames, implementing
heuristics based on the age and duration of reassembly holes, or some
combination.
12.5. Stream Commitment Attack
An adversarial endpoint can open lots of streams, exhausting state on
an endpoint. The adversarial endpoint could repeat the process on a
large number of connections, in a manner similar to SYN flooding
attacks in TCP.
Normally, clients will open streams sequentially, as explained in
Section 9.1. However, when several streams are initiated at short
intervals, transmission error may cause STREAM DATA frames opening
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streams to be received out of sequence. A receiver is obligated to
open intervening streams if a higher-numbered stream ID is received.
Thus, on a new connection, opening stream 2000001 opens 1 million
streams, as required by the specification.
The number of active streams is limited by the concurrent stream
limit transport parameter, as explained in Section 9.4. If chosen
judisciously, this limit mitigates the effect of the stream
commitment attack. However, setting the limit too low could affect
performance when applications expect to open large number of streams.
13. IANA Considerations13.1. QUIC Transport Parameter Registry
IANA [SHALL add/has added] a registry for "QUIC Transport Parameters"
under a "QUIC Protocol" heading.
The "QUIC Transport Parameters" registry governs a 16-bit space.
This space is split into two spaces that are governed by different
policies. Values with the first byte in the range 0x00 to 0xfe (in
hexadecimal) are assigned via the Specification Required policy
[RFC8126]. Values with the first byte 0xff are reserved for Private
Use [RFC8126].
Registrations MUST include the following fields:
Value: The numeric value of the assignment (registrations will be
between 0x0000 and 0xfeff).
Parameter Name: A short mnemonic for the parameter.
Specification: A reference to a publicly available specification for
the value.
The nominated expert(s) verify that a specification exists and is
readily accessible. The expert(s) are encouraged to be biased
towards approving registrations unless they are abusive, frivolous,
or actively harmful (not merely aesthetically displeasing, or
architecturally dubious).
The initial contents of this registry are shown in Table 7.
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